[asterisk-announce] Asterisk 17.0.0 Now Available

Asterisk Development Team asteriskteam at digium.com
Mon Oct 28 06:18:24 CDT 2019

The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
This release is available for immediate download at

The release of Asterisk 17.0.0 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:

Security bugs fixed in this release:
 * ASTERISK-28495 - res_pjsip_t38: 200 OK with SDP answer with
      declined stream causes crash
      (Reported by Alexei
 * ASTERISK-28447 - res_pjsip_messaging: In-dialog MESSAGE with
      no body causes crash
      (Reported by Gil Richard)
 * ASTERISK-28465 - Broken SDP can cause a segfault in a T.38
      (Reported by Francesco Castellano)
 * ASTERISK-28260 - Asterisk segfault when rtp negotiation is
      wrong or fails
      (Reported by Sotiris Ganouris)
 * ASTERISK-28127 - Buffer overflow for DNS SRV/NAPTR records
      (Reported by Jan Hoffmann)
 * ASTERISK-28013 - res_http_websocket: Crash when reading HTTP
      Upgrade requests
      (Reported by Sean Bright)

New Features made in this release:
 * ASTERISK-28403 - Add native Prometheus support to Asterisk
      (Reported by Matt Jordan)
 * ASTERISK-28375 - res_pjsip: New configuration setting to
      allow disabling norefersub
      (Reported by Dan Cropp)
 * ASTERISK-28320 - Added ARI resource
      (Reported by
      sungtae kim)
 * ASTERISK-28267 - res_stasis: Add ability to switch
      (Reported by Benjamin Keith Ford)
 * ASTERISK-28087 - add flag to allow CALLERID(num) to be placed
      in Contact header in chan_pjsip
      (Reported by Torrey
 * ASTERISK-27971 - res_pjsip: Implement additional SIP RFCs for
      Google Voice trunk compatability
      (Reported by Nick French)

Bugs fixed in this release:
 * ASTERISK-28561 - Asterisk Deadlocks
      (Reported by
 * ASTERISK-28575 - MWI Send Notify Crash on 16.6
      (Reported by Joshua Elson)
 * ASTERISK-28574 - pjproject fails to build on 16.6.0, works on
      (Reported by Niklas Larsson)
 * ASTERISK-28521 - pjsip: Memory Leak
      (Reported by Mark)
 * ASTERISK-28523 - Asterisk 16.5.0 Memory leak
      by Cyril Rami��re)
 * ASTERISK-28538 - chan_pjsip: Deadlock on fax detection
      (Reported by Joshua C. Colp)
 * ASTERISK-28536 - Asterisk release candidates fail to build on
      (Reported by Guido Falsi)
 * ASTERISK-28499 - translate: Crash when frame does not have a
      "src" field set
      (Reported by Gregory Massel)
 * ASTERISK-28488 - pjsip mwi: n+1 sip notify's sent on
      (Reported by Chris Savinovich)
 * ASTERISK-28509 - PJSIP cnonce generated on Linux contains 36
      characters, NEC only supports up to 32 characters
      (Reported by Dan Cropp)
 * ASTERISK-28505 - app_voicemail/IMAP: segfault in
      leave_voicemail because not checking mailstream
      by Alexei Gradinari)
 * ASTERISK-28487 - compile menuselect on gentoo
      by Kilburn)
 * ASTERISK-28472 - Asterisk occasionally passes a NULL as
      srtp->session to srtp_protect/unprotect causing SEGV
      (Reported by Jonas Swiatek)
 * ASTERISK-28498 - cel / cdr: Event times may be incorrect
      (Reported by Joshua C. Colp)
 * ASTERISK-28480 - json integer overflow in ssrc and timestamp

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

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

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

      (Reported by sungtae kim)
 * ASTERISK-28306 - res_pjsip_mwi: MWI NOTIFY occasionally takes
      minutes to be sent
      (Reported by Jared Hull)
 * ASTERISK-28332 - Variable ALTCONF ignored when service is
      used in Debian
      (Reported by Cirillo Ferreira)
 * ASTERISK-27964 - app_queue: ring_entry accesses nativeformats
      without channel lock or reference
      (Reported by Francisco
 * ASTERISK-28335 - stasis: Make topic and maybe subscription
      names unique and more useful
      (Reported by Joshua C. Colp)
 * ASTERISK-28321 - res_rtp_asterisk: Fixing possible divide by
      zero for rtcp stat calculation
      (Reported by sungtae kim)
 * ASTERISK-28322 - chan_pjsip: Add option to allow ignoring of
      183 without SDP
      (Reported by Torrey Searle)
 * ASTERISK-28328 - MeetMe global non-admin mute is muting
      admins that subsequently join
      (Reported by Philip Mott)
 * ASTERISK-28168 - app_queue: Adding a blank entry into sql
      queue_members crashes asterisk.
      (Reported by Michael)
 * ASTERISK-28323 - pjsip: sip.conf to pjsip.conf conversion
      script fails
      (Reported by Guido Weckwerth)
 * ASTERISK-28272 - The basic-pbx config samples don't produce a
      running asterisk
      (Reported by George Joseph)
 * ASTERISK-28312 - res_pjsip_diversion: Corrupted SIP Diversion
      field after handling a 302 redirect
      (Reported by Alex
 * ASTERISK-24173 - File menuselect/menuselect_gtk.c has no
      license header
      (Reported by Jeremy Lain��)
 * ASTERISK-28166 - app_voicemail: Asterisk unresponsive after
      changing voicemail password with ODBC
      (Reported by
 * ASTERISK-28309 - res_pjsip: Wrong Contact and Via fields with
      multiple UDP interfaces
      (Reported by Nikolay shakin)
 * ASTERISK-27992 - PJSIP: Adding `sends_registrations = yes` to
      pjsip_wizard.conf  causes crash
      (Reported by Jonathan
 * ASTERISK-28213 - res_pjsip: Threads pile up needlessly when
      AOR is blocked
      (Reported by Ross Beer)
 * ASTERISK-28301 - Allow voicemail boxes to be subscribed to
      with a presence event package
      (Reported by George Joseph)
 * ASTERISK-28303 - res_rtp_asterisk: Interaction between
      smoother and DTMF can cause out of order timestamps
      (Reported by Torrey Searle)
 * ASTERISK-28302 - ARI: "Error destroying mutex" when listing
      all ARI applications
      (Reported by Stefan Repke)
 * ASTERISK-28300 - AST_PBX_MAX_STACK is too low for some
      (Reported by George Joseph)
 * ASTERISK-28106 - Astricon Feedback: Unable to filter ARI
      events when GETting causes overload of events
      (Reported by
      George Joseph)
 * ASTERISK-28284 - switching between native_bridge and
      simple_bridge can cause one way audio
      (Reported by Torrey
 * ASTERISK-28251 - CI: Fix CI so it reverifies commit message
      (Reported by George Joseph)
 * ASTERISK-28277 - database: Add some basic logging
      (Reported by Joshua C. Colp)
 * ASTERISK-28181 - ari: Originating overwrites channel start
      (Reported by sungtae kim)
 * ASTERISK-28173 - Deadlock in chan_sip handling subscribe
      request during res_parking reload
      (Reported by Giuseppe
 * ASTERISK-28104 - AstriCon Feedback:  Automatically create a 1
      line dialplan context for stasis apps
      (Reported by George
 * ASTERISK-28271 - Opensuse Leap 15 --with-jannson-bundled will
      not compile
      (Reported by David Wilcox)
 * ASTERISK-28238 - PJSIP realtime. getcontext not working with
      (Reported by Ray)
 * ASTERISK-28263 - codec_opus: errors setting max_playback_rate
      and bitrate to "sdp"
      (Reported by Gianluca Merlo)
 * ASTERISK-28257 - res_http_websocket: PING / PONG opcodes
      break data reception
      (Reported by Jeremy Lain��)
 * ASTERISK-28250 - build: Cross-compilation fails for target
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-28252 - HangupHandler manager events are never
      (Reported by Gerald Schnabel)
 * ASTERISK-28231 - res_http_websocket: Not responding to
      Connection Close Frame (opcode 8)
      (Reported by Jeremy
 * ASTERISK-28249 - res_monitor: Segfault with
      (Reported by Valentin Vidi��)
 * ASTERISK-28244 - stasis: Filter messages at publishing to
      (Reported by Joshua C. Colp)
 * ASTERISK-28197 - stasis: ast_endpoint struct holds the
      channel_ids of channels past destruction in certain cases
      (Reported by Mohit Dhiman)
 * ASTERISK-28230 - res_rtp_asterisk: abs-send-time extension
      added with Asterisk 15.5.0 breaks GXV3140 video telephony
      (Reported by David Kuehling)
 * ASTERISK-28232 - core: RAII using clang use-after-scope
      (Reported by Diederik de Groot)
 * ASTERISK-28162 - [patch] need to reset DTMF last sequence
      number and timestamp on RTP renegotiation
      (Reported by
      Alexei Gradinari)
 * ASTERISK-28225 - app_voicemail: Channel variable
      VM_MESSAGEFILE not updated correctly if message marked "urgent"

      (Reported by boatright)
 * ASTERISK-28218 - app_queue: Asterisk crashes when using Queue
      with a pre-dial handler (option b)
      (Reported by Mark)
 * ASTERISK-28212 - stasis: Statistics broke ABI under developer
      (Reported by Joshua C. Colp)
 * ASTERISK-28222 - Regression: MWI polling no longer works
      (Reported by abelbeck)
 * ASTERISK-28221 - Bug in ast_coredumper
      (Reported by
      Andrew Nagy)
 * ASTERISK-28215 - app_voicemail: Leaving voicemail sometimes
      doesn't trigger NOTIFYs
      (Reported by George Joseph)
 * ASTERISK-27959 - [patch] Asterisk 15.4.1 h264 fmtp
      negotiation problem
      (Reported by David Kuehling)
 * ASTERISK-28201 - [patch] confbridge: no announce to the
      marked users when they join an empty conference
      by Alexei Gradinari)
 * ASTERISK-28117 - stasis: Add statistics for usage when in
      developer mode
      (Reported by Joshua C. Colp)
 * ASTERISK-28186 - stasis: Filter messages at publishing based
      on to_* presence
      (Reported by Joshua C. Colp)
 * ASTERISK-28194 - chan_sip: Leak using contact ACL
      (Reported by Giuseppe Sucameli)
 * ASTERISK-28157 - Asterisk crashes when the res_pjsip_*
      modules unload
      (Reported by sungtae kim)
 * ASTERISK-28125 - app_queue: Revert broken queue channel
      reference patch
      (Reported by lvl)
 * ASTERISK-27095 - chan_pjsip: When connected_line_method is
      set to invite, we're not trying UPDATE
      (Reported by George
 * ASTERISK-28182 - chan_pjsip: When connected_line_method is
      set to invite, asterisk is not trying UPDATE
      (Reported by
 * ASTERISK-28151 - app_voicemail: MWI fails with
      mailboxes=##@device instead of mailboxes=##@default
      (Reported by Ronald Raikes)
 * ASTERISK-28119 - stasis: Segment channel snapshot to reduce
      creation cost
      (Reported by Joshua C. Colp)
 * ASTERISK-28102 - stasis: Use implementation specific cache
      for channel snapshots
      (Reported by Joshua C. Colp)
 * ASTERISK-28159 - SIGABRT caused by stack corruption in
      hashkeys_read when no matching keys present
      (Reported by
      Michael Walton)
 * ASTERISK-28140 - repeated segmentation faults 
      (Reported by Eyal Hasson)
 * ASTERISK-28103 - stasis: Filter messages at publishing to
      reduce work done
      (Reported by Joshua C. Colp)
 * ASTERISK-28169 - ARI /channels/create handler causes core
      (Reported by sungtae kim)
 * ASTERISK-28129 - Incorrect Behavior for rewrite_contact when
      Re-Invite omits routset
      (Reported by Torrey Searle)
 * ASTERISK-28158 - Some conditions prevent running of el_end,
      break the terminal.
      (Reported by Corey Farrell)
 * ASTERISK-28110 - rtp: Incorrect Packetization
      by Robert Cripps)
 * ASTERISK-28146 - pbx_config: Only the first [globals] section
      is processed.
      (Reported by Corey Farrell)
 * ASTERISK-28150 - Formatting error in documentation
      (Reported by Scott Griepentrog)
 * ASTERISK-28081 - chan_sip: Asterisk 12+ chan_sip doesn't
      report AST_CEL_PICKUP in handle_invite_replaces
      by Luit van Drongelen)
 * ASTERISK-28137 - res_pjsip_notify: improve realtime
      performance on CLI completion on the endpoint
      (Reported by
      Alexei Gradinari)
 * ASTERISK-27980 - Caller ID cannot be changed on Attended
      Transfer before dialing out
      (Reported by Alexei Gradinari)
 * ASTERISK-28107 - app_confbridge:  Participant info labels
      aren't being added to the SDPs
      (Reported by George Joseph)
 * ASTERISK-28089 - function ast_sendtext() create RTP realtime
      packets with a trailing null byte in the payload
      by Emmanuel BUU)
 * ASTERISK-28076 - bridging: Asterisk crashes when receiving an
      empty realtime text frame
      (Reported by Emmanuel BUU)
 * ASTERISK-28084 - app_queue: QueueMemberStatus Event flooding
      (Reported by Andrej)
 * ASTERISK-28077 - res_pjsip: improve realtime performance on
      CLI 'pjsip show contacts'
      (Reported by Alexei Gradinari)
 * ASTERISK-27920 - app_queue: Queue member considered inuse
      after immediately hanging up during dialing.
      (Reported by
      Cao Minh Hiep)
 * ASTERISK-26094 - stasis: Playing MOH to bridge with ARI does
      not work
      (Reported by Cameron)
 * ASTERISK-28065 - res_odbc: missing SQL error diagnostic
      (Reported by Alexei Gradinari)
 * ASTERISK-28057 - chan_sip: SipNotify via AMI behaves
      differently to CLI
      (Reported by Peter Katzmann)
 * ASTERISK-28045 - configure script does not enforce
      libunbound2 version
      (Reported by Samuel Galarneau)
 * ASTERISK-28070 - testsuite: Sniffer assumes pjmedia will use
      ports below 10000
      (Reported by Joshua C. Colp)
 * ASTERISK-27854 - rtp: Crash in off-nominal case where RTP
      instance can't be set up
      (Reported by Lei Fu)
 * ASTERISK-28034 - chan_sip unstable with TLS after asterisk
      start or reloads
      (Reported by David Hajek)
 * ASTERISK-28059 - PJSIP: Update bundled PJPROJECT to version
      (Reported by Joshua C. Colp)
 * ASTERISK-28047 - chan_pjsip: Declined video stream is added
      when no video codecs configured and session refresh with removed
      video stream occurs
      (Reported by Will)
 * ASTERISK-28033 - AMI event "NewExten" is set to the wrong
      (Reported by lvl)
 * ASTERISK-28049 - res_pjproject build failure
      by Jaco Kroon)
 * ASTERISK-28029 - [patch] res_musiconhold : music on hold will
      not start if previous hold just reached end of file
      (Reported by Frederic LE FOLL)
 * ASTERISK-28005 - channel.c: ARI ring only once
      (Reported by Hajek Michal)
 * ASTERISK-28032 - Realtime queuemembers are not updated during
      retry phase
      (Reported by lvl)
 * ASTERISK-27988 - alembic: PJSIP
      "mwi_subscribe_replaces_unsolicited" field is integer not
      (Reported by Joshua C. Colp)
 * ASTERISK-28020 - res_pjsip_transport_websocket: Properly set
      'received' for IPv6
      (Reported by Sean Bright)
 * ASTERISK-28002 - When T.140 realtime text is negociated, a
      lot of debug traces are generated
      (Reported by Emmanuel
 * ASTERISK-27881 - PBX calls via chan_sip TCP trunk now get
      authentification error
      (Reported by Ian Gilmour)
 * ASTERISK-28022 - res_pjsip realtime: uri column in
      ps_contacts table can be too short
      (Reported by Florian
 * ASTERISK-27944 - res_pjsip_t38: Crash receiving 1xx responses
      other than 100 before 200 for T.38 reINVITE
      (Reported by
      Joshua Elson)
 * ASTERISK-28007 - rtcp-mux is put in SDP answer regardless of
      (Reported by Torrey Searle)
 * ASTERISK-27398 - No joint capabilities with video and
      audio-only streams
      (Reported by Benjamin Keith Ford)
 * ASTERISK-27973 - app_queue: QUEUESTATUS = CONTINUE instead
      (Reported by Valentin Safonov)
 * ASTERISK-27997 - pjproject_bundled: Fix for Solaris builds.
      Do not undef s_addr.
      (Reported by Alexander Traud)
 * ASTERISK-27999 - Wrong SRTP use status report
      by Salah Ahmed)
 * ASTERISK-28001 - res_pjsip_registrar: Improve performance of
      inbound handling
      (Reported by Joshua C. Colp)
 * ASTERISK-27966 - pjsip: Race condition in 183 re transmission
      can result in a deadlock
      (Reported by Torrey Searle)
 * ASTERISK-15331 - make menuselect fails due to undefined
      symbols (initscr32, w32addch) in menuselect_curses.o
      (Reported by Majdi Bsoul)
 * ASTERISK-14935 - [regression] menuselect compilation failure
      on Solaris 10
      (Reported by Samuel Owens)
 * ASTERISK-12382 - menuselect compilation failure on Solaris 10
      / gcc 3.4.3
      (Reported by rleasure)
 * ASTERISK-9107 - menuselect compilation failure on Solaris
      (Reported by Bob Atkins)
 * ASTERISK-27991 - BuildSystem: Enable Jansson in Solaris 11.
      (Reported by Alexander Traud)
 * ASTERISK-27548 - res_pjsip_endpoint_identifier_ip only
      matches against "generic string" headers
      (Reported by
      George Joseph)
 * ASTERISK-27990 - res_rtp_asterisk: Requires OpenSSL in
      Developer Mode.
      (Reported by Alexander Traud)
 * ASTERISK-27591 - Frack errors in stasis.c and memory leakage

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

Improvements made in this release:
 * ASTERISK-28443 - app_voicemail: remove dependency on stasis
      (Reported by Kevin Harwell)
 * ASTERISK-28442 - stasis_state: Create a stasis module to
      cache last known state
      (Reported by Kevin Harwell)
 * ASTERISK-28385 - res_ari_channels: Added detail hangup code
      (Reported by sungtae kim)
 * ASTERISK-28234 - pbx_dundi: Add IPv4/IPv6 dual bind support
      for DUNDi
      (Reported by Kirsty Tyerman)
 * ASTERISK-28401 - app_confbridge: Add *_all remb behavior
      (Reported by Joshua C. Colp)
 * ASTERISK-28400 - res_rtp_asterisk / res_pjsip_sdp_rtp: Add
      support for transport-cc
      (Reported by Joshua C. Colp)
 * ASTERISK-28363 - Millisecond-resolution call stats including
      PDD in channel variables
      (Reported by Antoni Goldstein)
 * ASTERISK-28378 - Added detail subscriber/subscription info
      for stasis show app cli
      (Reported by sungtae kim)
 * ASTERISK-20207 - Asterisk should clear out any .lock files in
      the voice mail directory on startup.
      (Reported by Steven
 * ASTERISK-28111 - build: CHANGES/UPGRADE are irritating to
      work with.
      (Reported by Corey Farrell)
 * ASTERISK-28264 - Added topic_all container
      (Reported by
      sungtae kim)
 * ASTERISK-28343 - Added app_name, app_data to channel type
      (Reported by sungtae kim)
 * ASTERISK-28326 - ari: Added timestamp for some ari events.
      (Reported by sungtae kim)
 * ASTERISK-28317 - Add logical group at DAHDIChannel event and
      create "dahdi_group" at CHANNEL function
      (Reported by
      Cirillo Ferreira)
 * ASTERISK-28279 - Added creation timestamp for bridge
      (Reported by sungtae kim)
 * ASTERISK-27483 - Allow wrapuptime to be set for each queue
      (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-28055 - app_queue: Per-member wrapup time missing
      from AddQueueMember application
      (Reported by Niksa Baldun)
 * ASTERISK-28292 - Changed to show all channel stats including
      wrong media
      (Reported by sungtae kim)
 * ASTERISK-28253 - res_pjsip_session: Adding rtcp stats result
      into the session
      (Reported by sungtae kim)
 * ASTERISK-28246 - Support skipping on the g726 format
      (Reported by Eyal Hasson)
 * ASTERISK-28196 - bridge_softmix: Does not support WebRTC
      source with multi video tracks.
      (Reported by Xiemin Chen)
 * ASTERISK-28198 - res_ari: Add new hangup causes for ARI
      Channel DELETE command
      (Reported by Sebastian Damm)
 * ASTERISK-28144 - [patch] New function PJSIP_PARSE_URI to
      parse an URI and return a specified part of the URI
      (Reported by Alexei Gradinari)
 * ASTERISK-28136 - Allow the sip_to_pjsip script to be used in
      a pipe
      (Reported by Pascal Cadotte Michaud)
 * ASTERISK-28046 - Remove stale nonoptreq references
      (Reported by Walter Doekes)
 * ASTERISK-27164 - [patch] Add IPv6 Support for DUNDi
      (Reported by Adam Secombe)
 * ASTERISK-28006 - PJSIP: Missing
      "party=calling"/"party=called" in Remote-Party-ID
      (Reported by Eric Dantie)
 * ASTERISK-27995 - pjproject_bundled: Find shared libraries in
      root --with-ssl=PATH.
      (Reported by Alexander Traud)
 * ASTERISK-27993 - pjsip_wizard example gives wrong info about
      unsupported SRV records
      (Reported by Jonathan Harris)
 * ASTERISK-27970 - res_rtp_asterisk: T.140 packets containing
      backspace or end of line are merged with regular text and it
      causes some UA to break
      (Reported by Emmanuel BUU)

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

Thank you for your continued support of Asterisk!
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-announce/attachments/20191028/006434ae/attachment-0001.html>

More information about the asterisk-announce mailing list