[asterisk-announce] Asterisk 15.0.0 Now Available

Asterisk Development Team asteriskteam at digium.com
Tue Oct 3 07:27:31 CDT 2017

The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
This release is available for immediate download at

The release of Asterisk 15.0.0 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:

Improvements made in this release:
 * ASTERISK-26230 - [patch] res_pjsip_mwi: unsolicited mwi could
      block PJSIP taskprocessor on startup
      (Reported by Alexei
 * ASTERISK-27043 - Core/BuildSystem: Add defines to fix build
      with LibreSSL
      (Reported by Guido Falsi)
 * ASTERISK-27042 - Unpatched asterisk sources fail to build on
      FreeBSD due to missing crypt.h file
      (Reported by Guido
 * ASTERISK-26419 - audiohooks: Remove redundant codec
      translations when using audiohooks
      (Reported by Michael
 * ASTERISK-26976 - libsrtp-2.x.x support
      (Reported by
 * ASTERISK-27014 - configurable busy_timeout in sqlite
      (Reported by Marek Cervenka)
 * ASTERISK-26124 - res_agi: Set audio format for EAGI audio
      (Reported by John Fawcett)
 * ASTERISK-26088 - Investigate heavy memory utilization by
      (Reported by Richard Mudgett)
 * ASTERISK-26427 - res_hep_rtcp: Asterisk Master will report
      channel name with res_hep_rtcp when using chan_sip
      (Reported by Nir Simionovich (GreenfieldTech - Israel))
 * ASTERISK-26932 - [patch] SIP/SDP: No rtpmap for static RTP
      payload IDs
      (Reported by Alexander Traud)
 * ASTERISK-26864 - res_pjsip_session: Add support for overlap
      (Reported by Richard Begg)
 * ASTERISK-26846 - chan_sip: Add rtcp-mux support
      (Reported by Sean Bright)
 * ASTERISK-26568 - pbx_spool: OUTGOING_RETRY variable
      (Reported by Roman Shubovich)
 * ASTERISK-26292 - app_confbridge: 3D-Conferencing via Binaural
      (Reported by Dennis Guse)
 * ASTERISK-23828 - pjsip - Need a command to list active SIP
      (Reported by Rusty Newton)
 * ASTERISK-26559 - app_queue:  New service level calculation
      (Reported by Sebastian Gutierrez)
 * ASTERISK-26658 - Add ability for dialplan show to display
      filenames/line numbers of registered extensions
      by Jonathan R. Rose)
 * ASTERISK-26527 - Testsuite: increase timeout to check "core
      fullybooted wait" up to 30 sec
      (Reported by Badalian
 * ASTERISK-22992 - [patch]Asterisk app_originate doesn't allow
      setting Caller*ID on the originating channel
      (Reported by
      Anthony Messina)
 * ASTERISK-26624 - res_calendar_caldav: Add support for gmail
      (Reported by Eduardo Scudeller Libardi)
 * ASTERISK-26562 - app_controlplayback: Transmit Silence on
      ControlPlayback pause
      (Reported by Mikheili Dautashvili)
 * ASTERISK-24517 - TLS support for Solaris, Ming and non-glibc
      Linux systems
      (Reported by Timo Teräs)
 * ASTERISK-26540 - cdr_radius: use radcli instead of
      (Reported by Tzafrir Cohen)
 * ASTERISK-26558 - app_queue: add variable to know if the call
      is not answered after a queue
      (Reported by Sebastian
 * ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry
      (Reported by Sebastian Gutierrez)
 * ASTERISK-26217 - [patch] Codec 2 Mode 2400
      (Reported by
      Alexander Traud)
 * ASTERISK-26538 - codec_opus: Add sample to
      (Reported by Kevin
 * ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps',
      and 'ari set debug' CLI commands
      (Reported by Matt Jordan)
 * ASTERISK-26418 - res_rtp_asterisk: Speed up ICE resolution by
      blacklisting host subnets that are not involved in RTP
      (Reported by Michael Walton)
 * ASTERISK-26422 - [patch] Force calendars to do new fetch
      after module reload
      (Reported by Ludovic Gasc (Eyepea))
 * ASTERISK-26398 - core: Remove ABI differences of LOW_MEMORY
      (Reported by Corey Farrell)
 * ASTERISK-26409 - codec_opus: Update Asterisk to support the
      translation codec.
      (Reported by Kevin Harwell)
 * ASTERISK-26289 - Announcer channels in ConfBridges cause
      (Reported by Mark Michelson)
 * ASTERISK-26321 - ARI : Add reason answered_elsewhere to
      channel hangup
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-25980 - [patch]res_fax: set FAXMODE variable to let
      dialplan know what fax transport was used
      (Reported by
      Alexei Gradinari)
 * ASTERISK-26218 - [patch] iLBC 20
      (Reported by Alexander
 * ASTERISK-26190 - [patch] SRTP: Enable AES-256 and AES-GCM.
      (Reported by Alexander Traud)
 * ASTERISK-26220 - Add support for noreturn function
      (Reported by Corey Farrell)

Bugs fixed in this release:
 * ASTERISK-27152 - Sending a "tel" uri in a From or To header
      in an unauthenticated message causes asterisk to crash
      (Reported by Ross Beer)
 * ASTERISK-27103 - core: ast_safe_system command injection
      (Reported by Corey Farrell)
 * ASTERISK-27013 - res_rtp_asterisk: Media can be hijacked even
      with strict RTP enabled
      (Reported by Joshua Colp)
 * ASTERISK-27231 - res_rtp_asterisk: Allow remote SSRC to
      change due to renegotiation
      (Reported by Joshua Colp)
 * ASTERISK-27222 - core: Don't queue up multiple video update
      (Reported by Joshua Colp)
 * ASTERISK-27212 - bridge_softmix: Quickly joining/leaving may
      cause video stream to remain in SFU
      (Reported by Richard
 * ASTERISK-27202 - If wget is not installed and "or" is not
      available, external components (excluding pjsip) are not
      (Reported by Seán C. McCord)
 * ASTERISK-27200 - manager: hook event is not being raised
      (Reported by Kevin Harwell)
 * ASTERISK-27179 - res_pjsip_session: Handling of 'msid' is
      (Reported by Kevin Harwell)
 * ASTERISK-27182 - bridge: Crash when mapping streams
      (Reported by Joshua Colp)
 * ASTERISK-27189 - Make --with-pjproject-bundled the default
      for Asterisk 15
      (Reported by George Joseph)
 * ASTERISK-27180 - channel: requester leaks joint_cap on
      (Reported by Corey Farrell)
 * ASTERISK-27171 - Asterisk 15.0.0-Beta1 does not compile
      (Reported by Ira Emus)
 * ASTERISK-27119 - res_pjsip: parse/add msid attribute when
      webrtc is enabled
      (Reported by Kevin Harwell)
 * ASTERISK-27143 - bridge_softmix / res_rtp_asterisk: Fix
      packet loss and renegotiation issues.
      (Reported by Joshua
 * ASTERISK-27142 - sounds: Conflict between files in
      asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
      (Reported by Corey Farrell)
 * ASTERISK-25810 - say.c calls for sounds in the subdir
      "digits" that don't exist (in Core). SayUnixTime or other Say...
      apps will fail out when they call these sounds.
      by Nicolas Riendeau)
 * ASTERISK-27136 - bridge_softmix: Don't reorder SFU streams
      (Reported by Joshua Colp)
 * ASTERISK-27134 - bridge_softmix: Reuse any removed streams
      for video
      (Reported by Joshua Colp)
 * ASTERISK-27133 - res_rtp_asterisk: RTCP does not use ICE when
      RTCP-MUX in use
      (Reported by Joshua Colp)
 * ASTERISK-27123 - confbridge: Name recordings are left on
      (Reported by Sergej Kasumovic)
 * ASTERISK-27122 - chan_iax2: On reload MWI taskprocessors keep
      adding up
      (Reported by Sergej Kasumovic)
 * ASTERISK-26807 - sounds: New 3-D Binaural audio features
      require new sound prompts
      (Reported by Rusty Newton)
 * ASTERISK-25816 - French conf-adminmenu, conf-usermenu prompts
      differ in content from the English files
      (Reported by
      Benoit Duverger)
 * ASTERISK-26274 - Resolve open sounds issues and then create a
      new sounds release (1.5.1? or 1.6?)
      (Reported by Rusty
 * ASTERISK-27118 - res_pjsip_session / res_rtp_asterisk: Add
      support for BUNDLE
      (Reported by Joshua Colp)
 * ASTERISK-27036 - res_pjsip: Asterisk crashes when an
      extension tries to use PJSIP trunk with from_user containing
      (Reported by Maxim Vasilev)
 * ASTERISK-27023 - res_rtp_asterisk: Deadlock when TURN session
      in use
      (Reported by Jatin Jain)
 * ASTERISK-27106 - [patch] autodomain (SIP Domain Support): Add
      only really different domain with TLS.
      (Reported by
      Alexander Traud)
 * ASTERISK-27093 - ODBC deadlocks when app_directory tries to
      play back non-existent voicemail greeting
      (Reported by
      James Terhune)
 * ASTERISK-27100 - channel: ast_waitfordigit_full fails to
      clear flag in an error branch.
      (Reported by Corey Farrell)
 * ASTERISK-27090 - PJSIP: Deadlock using TCP transport
      (Reported by Richard Mudgett)
 * ASTERISK-26997 - Create an StreamEcho dialplan application
      (Reported by Kevin Harwell)
 * ASTERISK-27076 - chan_pjsip: Add support for multiple
      (Reported by Joshua Colp)
 * ASTERISK-27088 - res_rtp_asterisk: Better handle ICE
      renegotiation and unidirectional negotiation
      (Reported by
      Joshua Colp)
 * ASTERISK-26978 - rtp: Crash in ast_rtp_codecs_payload_code()

      (Reported by Ross Beer)
 * ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY
      (Reported by Ove Aursand)
 * ASTERISK-27065 - call hangup after leaving app_queue
      (Reported by Marek Cervenka)
 * ASTERISK-24052 - app_voicemail reloads result in leaked IMAP
      (Reported by Louis Jocelyn Paquet)
 * ASTERISK-27074 - core_local: local channel data not being
      properly unref'ed and unlocked
      (Reported by Kevin Harwell)
 * ASTERISK-27075 - bridge: stuck channel(s) after failed
      attended transfer
      (Reported by Kevin Harwell)
 * ASTERISK-27060 - Comment typo format_g729.c
      by Matthew Fredrickson)
 * ASTERISK-27041 - Core/PBX: [patch] Deadlock between dialplan
      execution and application unregistration
      (Reported by
      Frederic LE FOLL)
 * ASTERISK-25370 - res_corosync segfaults at startup with
      corosync version > 2.x
      (Reported by mdu113)
 * ASTERISK-27026 - res_ari: Crash when no ari.conf
      configuration file exists
      (Reported by Ronald Raikes)
 * ASTERISK-27016 - Crash occurs when a channel in a
      'mixing,dtmf_events' bridge is muted multiple times.
      (Reported by Chris Howard)
 * ASTERISK-27057 - Seg Fault in ast_sorcery_object_get_id at
      (Reported by Ryan Smith)
 * ASTERISK-27024 - nat/external_media settings ignored in
      (Reported by Christopher van de Sande)
 * ASTERISK-27022 - res_rtp_asterisk: Incorrect SSRC change for
      RTCP component
      (Reported by Michael Walton)
 * ASTERISK-26923 - bridging: T.38 request is lost when channels
      are added to bridge
      (Reported by Torrey Searle)
 * ASTERISK-27053 - res_pjsip_refer/session: Calls dropped
      during transfer
      (Reported by Kevin Harwell)
 * ASTERISK-27052 - Asterisk build process fails with flag
      --with-pjproject-bundled with curl download command and slow
      (Reported by alex)
 * ASTERISK-27046 - res_pjsip_transport_websocket: segfault in
      (Reported by Jørgen H)
 * ASTERISK-27039 - chan_pjsip: Device state is idle when
      channel from endpoint is in early media
      (Reported by
      Joshua Colp)
 * ASTERISK-26996 - chan_pjsip: Flipping between codecs
      (Reported by Michael Maier)
 * ASTERISK-26281 - chan_pjsip would send INVITE to
      'Unreachable' endpoints
      (Reported by Jacek Konieczny)
 * ASTERISK-26973 - bridge: Crash when freeing frame and
      (Reported by Michel R. Vaillancourt)
 * ASTERISK-19291 - Background in realtime
      (Reported by
      Andrew Nowrot)
 * ASTERISK-27025 - channel / meetme: Fix missing parentheses
      (Reported by Joshua Colp)
 * ASTERISK-27021 - GET /recordings/stored returns 500 Internal
      Server Error
      (Reported by Tim Morgan)
 * ASTERISK-24858 - [patch]Asterisk 13 PJSIP sends RTP packets
      in wrong byte order on Intel platform when using slin codec
      (Reported by Frankie Chin)
 * ASTERISK-23951 -  Asterisk attempts and fails to build
      format_mp3 even if mp3lib was not downloaded
      (Reported by
      Tzafrir Cohen)
 * ASTERISK-25294 - srtp's crypto_get_random deprecated
      (Reported by Tzafrir Cohen)
 * ASTERISK-23839 - AGI - RECORD FILE - documentation doesn't
      describe BEEP argument
      (Reported by Rusty Newton)
 * ASTERISK-22432 - Async AGI crashes Asterisk when issuing "set
      variable" command without args
      (Reported by Antoine
 * ASTERISK-25662 - Malformed AGI 520 Usage response
      (Reported by Tony Mountifield)
 * ASTERISK-27008 - res_format_attr_h264: SDP parse fails if
      fmtp optional parameters have a space
      (Reported by John
 * ASTERISK-26399 - app_queue: Agent not called when caller is
      (Reported by wushumasters)
 * ASTERISK-26400 - app_queue: Queue member stops being called
      after AMI "Redirect" action for queues with wrapuptime
      (Reported by Etienne Lessard)
 * ASTERISK-26715 - app_queue: Member will not receive any new
      calls after doing a transfer if wrapuptime = greater than 0 and
      using Local channel
      (Reported by David Brillert)
 * ASTERISK-26975 - app_queue: Non-zero wrapup time can cause
      agents not to receive queue calls after transfer queue call
      (Reported by Lorne Gaetz)
 * ASTERISK-27012 - app_confbridge: ConfBridge sometimes does
      not play user name recording while leaving
      (Reported by
      Robert Mordec)
 * ASTERISK-26979 - res_rtp_asterisk: SRTP unprotect failed with
      authentication failure 10 or 110
      (Reported by Javier
      Riveros )
 * ASTERISK-26982 - chan_sip: rtcp_mux setting may cause ice
      completion failure/delay if client offers rtcp-mux as
      (Reported by Stefan Engström)
 * ASTERISK-26939 - Out of bound memory access in PJSIP
      multipart parser crashes Asterisk
      (Reported by Sandro
 * ASTERISK-26940 - Asterisk Skinny memory exhaustion
      vulnerability leads to DoS
      (Reported by Sandro Gauci)
 * ASTERISK-26938 - Heap overflow in CSEQ header parsing affects
      Asterisk chan_pjsip and PJSIP
      (Reported by Sandro Gauci)
 * ASTERISK-26789 - Audit manipulation of channel flags without
      (Reported by Joshua Colp)
 * ASTERISK-26998 - res_pjsip_session: INVITE retransmissions
      could still setup the same call again.
      (Reported by
      Richard Mudgett)
 * ASTERISK-26143 - res_rtp_asterisk: One way audio when
      (Reported by Henning Holtschneider)
 * ASTERISK-26333 - Problems with Blind Transfer, PJSIP (Aastra
      (Reported by Matthias Binder)
 * ASTERISK-26606 - tcptls: Incorrect OpenSSL function call
      leads to misleading error report
      (Reported by Bob Ham)
 * ASTERISK-26983 - Crash in Manager Reload when TLS Config
      (Reported by Joshua Elson)
 * ASTERISK-25032 - [patch]cel_odbc sometimes inserts CEL with
      wrong eventtime
      (Reported by Etienne Lessard)
 * ASTERISK-26173 - func_cdr: CDR function does not permit empty
      values to be assigned
      (Reported by gkloepfer)
 * ASTERISK-25506 - [patch]CONFBRIDGE failure after an
      app_confbrige.so module reload results in segfault or
      error/warning messages.
      (Reported by Frederic LE FOLL)
 * ASTERISK-24529 - Using AMI Action Bridge to on an already
      bridged channel causes the incorrect return priority to be used

      (Reported by Corey Farrell)
 * ASTERISK-26966 - bridge_simple: Add support for streams
      (Reported by Kevin Harwell)
 * ASTERISK-26974 - res_pjsip: Deadlock in T.38 framehook
      (Reported by Richard Mudgett)
 * ASTERISK-26908 - res_pjsip: The ChanIsAvail causes a
      res_pjsip session to be leaked.
      (Reported by Richard
 * ASTERISK-26959 - dial: Allow topology of dialing channel to
      influence dialed channel
      (Reported by Joshua Colp)
 * ASTERISK-25823 - SIGSEGV, Segmentation fault. -
      ../sysdeps/x86_64/strlen.S: No such file or directory.
      (Reported by Andreas Krüger)
 * ASTERISK-26926 - func_speex: Crash caused by frame with no
      (Reported by Richard Kenner)
 * ASTERISK-26964 - res_pjsip_session: Wrong From on reinvite
      when request and To URI differ
      (Reported by Yasin CANER)
 * ASTERISK-26930 - pjproject/Makefile.rules for pjsip 2.6 build
      fails for non-SSE2 instrunction Linux
      (Reported by
 * ASTERISK-26922 - chan_sip: tcpbind uses wrong source address

      (Reported by Ksenia)
 * ASTERISK-26929 - pjsip: Add database tables for RLS
      (Reported by Joshua Colp)
 * ASTERISK-26949 - sdp: Implement T.38
      (Reported by
      Joshua Colp)
 * ASTERISK-26953 - Asterisk crash if hep.conf have some missing
      (Reported by Joel Vandal)
 * ASTERISK-26890 - STUN server with non-default-route transport
      causes INVITE delay
      (Reported by George Joseph)
 * ASTERISK-26951 - chan_sip: ACK with SDP does not update a
      direct media bridge
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-26692 - res_rtp_asterisk: Crash in
      dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
      (Reported by Sebastian Gutierrez)
 * ASTERISK-26835 - res_rtp_asterisk: Crash when freeing RTCP
      address string
      (Reported by Niklas Larsson)
 * ASTERISK-26853 - res_rtp_asterisk: Crash in pjnath when
      receiving packet
      (Reported by Adagio)
 * ASTERISK-26613 - format_wav: wav16 format read file only by
      320 - half of frame
      (Reported by Vitaly K)
 * ASTERISK-26169 - format_ogg_vorbis: Memory leak using OGG in
      (Reported by Ivan Myalkin)
 * ASTERISK-21856 - STUN never works when asterisk started
      without internet access
      (Reported by Jeremy Kister)
 * ASTERISK-20984 - Audible clicks when playing sox encoded au
      file with STREAM FILE AGI command
      (Reported by Roman S.)
 * ASTERISK-26528 - [UBSAN] strings.h:signed integer overflow in
      (Reported by Badalian Vyacheslav)
 * ASTERISK-26851 - res_pjsip_sdp_rtp: RTP instance does not use
      same IP as explicit transport
      (Reported by Richard Begg)
 * ASTERISK-26903 - Listening TCP/TLS sockets stop when
      temporarily out of open files
      (Reported by Walter Doekes)
 * ASTERISK-26928 - pjsip: Add database tables for PUBLISH
      (Reported by Joshua Colp)
 * ASTERISK-26927 - pjproject_bundled: Crash on
      pj_ssl_get_info() while ioqueue_on_read_complete().
      (Reported by Alexander Traud)
 * ASTERISK-26905 - pjproject_bundled:  Merge 3 upstream
      deadlock patches into bundled
      (Reported by Ross Beer)
 * ASTERISK-26920 - app_queue: PAUSEALL/UNPAUSEALL does not log
      (Reported by Troy Bowman)
 * ASTERISK-26897 - chan_sip: Security vulnerability with client
      code header
      (Reported by Alex Villacís Lasso)
 * ASTERISK-25974 - Unused realtime MOH classes not purged on
      'moh reload'
      (Reported by Sébastien Couture)
 * ASTERISK-26916 - res_pjsip: Excessive refcount reached on
      transport ao2 object
      (Reported by Ross Beer)
 * ASTERISK-21721 - SIP Failed to parse multiple Supported:
      (Reported by Olle Johansson)
 * ASTERISK-26915 - chan_sip: Session Timers required but
      refused wrongly.
      (Reported by Alexander Traud)
 * ASTERISK-26363 - res_pjsip: Bye sent to sip trunk is not
      authenticated even after receiving a 407 error code
      (Reported by Yaacov Akiba Slama)
 * ASTERISK-26896 - Overflow of buffer to PQEscapeStringConn
      with large app_args causes ABRT
      (Reported by twisted)
 * ASTERISK-26705 - libasteriskssl.so not found when asterisk is
      installed for the 1st time
      (Reported by George Joseph)
 * ASTERISK-26900 - sdp: Add support for connection address
      management and topology updating
      (Reported by Joshua Colp)
 * ASTERISK-21009 - xmpp_pubsub_unsubscribe: Could not create IQ
      when creating pubsub unsubscription on client
      (Reported by
      Marcello Ceschia)
 * ASTERISK-25490 - [patch]SDP crypto tag is validated
      (Reported by Joerg Sonnenberger)
 * ASTERISK-26885 - channel: Support dynamic number of file
      (Reported by Joshua Colp)
 * ASTERISK-26086 - res_musiconhold: format option is not
      documented adequately
      (Reported by Jens Bürger)
 * ASTERISK-23996 - No core dumps because of res_musiconhold
      (Reported by Walter Doekes)
 * ASTERISK-24712 - xmpp: starttls problem causes connection
      (Reported by Matthias Urlichs)
 * ASTERISK-26814 - pjproject_bundled build fails to download
      pjproject source when using cURL
      (Reported by Gergely
 * ASTERISK-23510 - JABBER_STATUS fails with improper code 7 for
      unavailable clients
      (Reported by Anthony Critelli)
 * ASTERISK-21855 - Asterisk crashes when XMPP message is sent
      (JabberSend) and no internet connection is available
      (Reported by Jeremy Kister)
 * ASTERISK-25622 - WARNING for "JABBER: socket read error"
      should be more specific
      (Reported by Sean Darcy)
 * ASTERISK-26515 - rtp_engine: Allocate RTP payloads on a
      per-session basis
      (Reported by Joshua Colp)
 * ASTERISK-26818 - cdr: Problem setting variables in h exten
      (Reported by Sebastian Gutierrez)
 * ASTERISK-26850 - res_hep_pjsip: Asterisk insert wrong
      protocol name in "Protocol ID" field in HEP packets
      (Reported by Max Norba)
 * ASTERISK-26484 - res_pjsip_messaging: Crash when using
      invalid URI in MessageSend 'from' argument.
      (Reported by
      Vinod Dharashive)
 * ASTERISK-26776 - res_pjsip_pubsub: Crash when generating
      xpidf content
      (Reported by Andrew Green)
 * ASTERISK-26880 - Asterisk crashes when multiple speex users
      join confbridge with pp_vad and dtx enabled
      (Reported by
      Kirsty Tyerman)
 * ASTERISK-26875 - app_mixmonitor: Recording out of sync when
      183 but no RTP
      (Reported by Aaron An)
 * ASTERISK-26862 - app_queue: Queue stops calling members with
      local interface after forwarding in previous call
      (Reported by Robert Mordec)
 * ASTERISK-26732 - res_rtp_asterisk: Implement RTCP
      Multiplexing - breaking WebRTC in Chrome
      (Reported by Dan
 * ASTERISK-26879 - PJSIP external_media_address ignored if no
      local_net options are provided
      (Reported by Matt Jordan)
 * ASTERISK-26867 - autochan: Locking in a function
      ast_autochan_destroy() on destroyed channel (after masquerade).

      (Reported by Krzysztof Trempala)
 * ASTERISK-26869 - res_pjsip_refer: blind call transfer w/o a
      user name doesn't go to the s extension
      (Reported by
      Torrey Searle)
 * ASTERISK-26668 - core: Malformed pattern matching extension
      (various factors) results in crash
      (Reported by xrobau)
 * ASTERISK-26865 - chan_iax2: Reload of iax peer results in
      loss of host address/port
      (Reported by Richard Begg)
 * ASTERISK-26872 - Bundled pjproject fails to build when
      tarball downloaded with curl due to md5 verification failure in
      Docker containers (or when there is no terminal)
      by Matt Jordan)
 * ASTERISK-26717 - Document the fact that Asterisk HEP support
      only works with the PJSIP channel driver
      (Reported by
      Olivier Krief)
 * ASTERISK-26643 - Extra new line in Device field of
      DeviceStateChange AMI Event after restart of Asterisk
      (Reported by Roman Bedros)
 * ASTERISK-25237 -  stasis_cache.c:845 caching_topic_exec: -
      misleading ERROR message
      (Reported by Smirnov Aleksey)
 * ASTERISK-26857 - chan_pjsip: Dialplan function race
      (Reported by Joshua Colp)
 * ASTERISK-26822 - pjsip/cli_commands: pjsip show channelstats
      shows wrong codec
      (Reported by Kevin Harwell)
 * ASTERISK-26353 - res_musiconhold: musiconhold seems to think
      that the general section is a class and issues warning
      (Reported by Jonathan Harris)
 * ASTERISK-26685 - res_pjsip: Crash when using IPv6 and
      Transport ws,wss
      (Reported by Michael Balen)
 * ASTERISK-24562 - app_voicemail: Cannot set fromstring on a
      per-mailbox basis
      (Reported by Mark Scholten)
 * ASTERISK-26842 - Websocket becomes disconnected when trying
      to place call from browser
      (Reported by Mark Michelson)
 * ASTERISK-26841 - chan_sip: Call not cancelled after receiving
      a 422 response
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-26839 - core: Implement stream topology changing in
      (Reported by Joshua Colp)
 * ASTERISK-26598 - Saynumber is trying to get "and" from
      "digits/" subfolder
      (Reported by Jonathan Harris)
 * ASTERISK-17067 - Long lines in call files cause spurious
      syntax error
      (Reported by Dave Olszewski)
 * ASTERISK-26796 - res_pjsip_transport_websocket: Via header is
      'WS' when it should be 'WSS'
      (Reported by Jørgen H)
 * ASTERISK-26816 - Implement ast_read_stream in channels
      (Reported by Joshua Colp)
 * ASTERISK-25628 - res_config_pgsql: should match the behavior
      of other drivers so that queue_log can disable adaptive logging

      (Reported by Dmitry Wagin)
 * ASTERISK-26774 - core: Playback URL fails after some time
      (Reported by Igor Gamayunov)
 * ASTERISK-26825 - pjsip.conf.sample: user_agent: still refers
      to branch 12
      (Reported by Tzafrir Cohen)
 * ASTERISK-26823 - PJSIP: Persistent subscriptions can cause
      FRACKs if endpoint does not exist
      (Reported by Mark
 * ASTERISK-26623 - res_pjsip: Crash when calling
      (Reported by Jørgen H)
 * ASTERISK-26808 - res_pjsip_outbound_registration doesn't know
      about network change events
      (Reported by George Joseph)
 * ASTERISK-26781 - bridge: Passing the 'p' (play tone) flag to
      Bridge() application results in garbled audio
      (Reported by
      Sean Bright)
 * ASTERISK-26782 - res_pjsip: URI requirement for fields is not
      consistently documented and error does not provide indication
      (Reported by Peter Sokolov)
 * ASTERISK-26793 - Implement ast_write_stream in channels
      (Reported by George Joseph)
 * ASTERISK-26812 - [patch] Fix download_externals To Allow The
      Use Of curl Or wget
      (Reported by Michael L. Young)
 * ASTERISK-18271 - Pattern matching with res_config_mysql
      extensions does not behave as expected
      (Reported by
      Charlie Smurthwaite)
 * ASTERISK-26811 - stream: Add streams to "core show channel"
      (Reported by Joshua Colp)
 * ASTERISK-18731 - [patch] DUNDi weight parameter not processed
      (Reported by Peter Racz)
 * ASTERISK-26799 - res_pjsip: Using an auth object for inbound
      and outbound authentication fails.
      (Reported by Richard
 * ASTERISK-26669 - PJSIP Segfault 13.13.1 (Bundled PJSIP)
      (Reported by Nic Colledge)
 * ASTERISK-26738 - Frequent segfaults since activation of DNS
      SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c,
      and pj_atomic_inc_and_get at pj/os_core_unix.c
      by Michael Maier)
 * ASTERISK-25893 - Function vmauthenticate accesses
      uninitialized memory
      (Reported by Filip Jenicek)
 * ASTERISK-26580 - [patch] Error during LDAP modify action when
      user unregisters
      (Reported by Nicholas John Koch)
 * ASTERISK-26802 - [patch] Integrity Check Of PJSIP Download
      (Reported by Michael L. Young)
 * ASTERISK-15858 - [patch] Fix query with double backslash in
      string literals and stop log warnings
      (Reported by
      Humberto Figuera)
 * ASTERISK-26057 - res_config_sqlite3 uses incorrect query -
      unnecessary escape
      (Reported by Stepan)
 * ASTERISK-23457 - SQlite3: Realtime queue loading fails after
      PRAGMA query result
      (Reported by Scott Griepentrog)
 * ASTERISK-26794 - http: Crash on Reload Only in
      (Reported by Joshua Elson)
 * ASTERISK-26714 - Phone default have not ringing on ARM
      (Reported by Igor Goncharovsky)
 * ASTERISK-26696 - pjsip_pubsub: PJSIP Subscription Persistence
      in AstDB Does not update on subscription refresh
      by Zach R)
 * ASTERISK-26756 - res_pjsip_mwi: Asterisk does not terminate
      MWI subscription
      (Reported by Carl Fortin)
 * ASTERISK-26790 - Implement stream topology (non-change
      request) API usage in channels
      (Reported by George Joseph)
 * ASTERISK-26723 - VoiceMailPlayMsg not playing messages via
      (Reported by Ryan Rittgarn)
 * ASTERISK-18286 - [patch] 'Silence' is truncated in Record()
      (Reported by var)
 * ASTERISK-26775 - app_queue: reset abandoned in service level

      (Reported by Sebastian Gutierrez)
 * ASTERISK-26786 - Implement ast_stream_topology API
      (Reported by George Joseph)
 * ASTERISK-26248 - chan_pjsip: Error when calling PJSIP client
      with domain specified
      (Reported by Norbert Varga)
 * ASTERISK-26788 - core: Protect flags during ast_waitfor
      (Reported by Joshua Colp)
 * ASTERISK-26115 - pbx: AMI Originate ignore "failed" extension
      on call failure
      (Reported by Nasir Iqbal)
 * ASTERISK-26773 - stream: Add basic API
      (Reported by
      Joshua Colp)
 * ASTERISK-26785 - configs/samples:  The 'identify' entry is in
      the wrong section in sorcery.conf.sample
      (Reported by
      Torrey Searle)
 * ASTERISK-26772 - Crash in srv.c on startup with pjsip
      (Reported by nappsoft)
 * ASTERISK-26770 - res_stasis_device_state: Duplicate
      subscriptions when multiple received at same time
      (Reported by Joshua Colp)
 * ASTERISK-26767 - ARI channelvars cause memory leak
      (Reported by Sébastien Duthil)
 * ASTERISK-26716 - ari: Channels with pre-dial handlers cannot
      be hung up via ARI
      (Reported by Tom Pawelek)
 * ASTERISK-26632 - core: Possibility of a frame "imbalance"
      leading to stuck channels.
      (Reported by Mark Michelson)
 * ASTERISK-25951 - res_agi:  run_agi eats frames it shouldn't
      (Reported by George Joseph)
 * ASTERISK-26343 - ASTERISK-25951 causes issues for callerid
      manipulation through agi
      (Reported by Morten Tryfoss)
 * ASTERISK-26704 - res_odbc.conf contains deprecated
      configuration: 'pooling', 'shared_connections', 'limit', and
      'idlecheck' options were replaced by 'max_connections'.
      (Reported by Anthony Messina)
 * ASTERISK-26765 - res_resolver_unbound: FRACK! Excessive ref
      count trap tripped.
      (Reported by Richard Mudgett)
 * ASTERISK-21094 - MixMonitorMute mutes through stream if
      already slinear (e.g. Originate)
      (Reported by David
 * ASTERISK-26679 - Crash on invalid contact domain (pjsip aor)

      (Reported by Dmitriy)
 * ASTERISK-26699 - res_pjsip: Assertion when sending OPTIONS
      request  to endpoint
      (Reported by Ross Beer)
 * ASTERISK-26754 - build_tools: make_build_h does not handle \
      in user name 
      (Reported by Kirill Katsnelson)
 * ASTERISK-26755 - app_queue: Random queues disappear on "core
      reload queue all"
      (Reported by Kirill Katsnelson)
 * ASTERISK-26735 - res_pjsip_endpoint_identifier_ip:
      "srv_lookups" after match in .conf has no effect
      by Michael Maier)
 * ASTERISK-26693 - res_pjsip_endpoint_identifier_ip: Add
      support for SRV
      (Reported by Joshua Colp)
 * ASTERISK-26743 - PJPROJECT: Detecting compiled max log level
      does not work.
      (Reported by Richard Mudgett)
 * ASTERISK-26731 - res_sorcery_memory_cache: memory leak on
      every sorcery memory cache populate
      (Reported by Ustinov
 * ASTERISK-26739 - voicemail API test: confuses expected and
      actual values
      (Reported by Tzafrir Cohen)
 * ASTERISK-26740 - voicemail API test: uses varlibdir instead
      of datadir for a sound file
      (Reported by Tzafrir Cohen)
 * ASTERISK-26665 - app_queue: Agent ringing, Caller hangup
      before timeout, no agent name logged - missing RINGNOANSWER?
      (Reported by Marek Cervenka)
 * ASTERISK-26710 - [patch] res_rtp_asterisk: CHANNEL arguments,
      (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return
      (Reported by Aaron An)
 * ASTERISK-26672 - Crash when setting remote address on RTP
      (Reported by Richard Mudgett)
 * ASTERISK-26670 - [patch] Outgoing SIP-URI Dialing via PJSIP
      (Reported by Alexander Traud)
 * ASTERISK-26691 - Remember SDP negotiation on
      (Reported by Alexander Traud)
 * ASTERISK-26673 - chan_pjsip: Crash when using CHANNEL
      dialplan function around masquerade
      (Reported by Joshua
 * ASTERISK-26684 - res_pjsip: Various issues with compact SIP
      (Reported by Joshua Elson)
 * ASTERISK-26683 - res_calendar: Calendars duplicated after
      module reload
      (Reported by Martin Tomec)
 * ASTERISK-26655 - [patch]pjsip: Transfers Broken with Compact
      Headers Enabled
      (Reported by JoshE)
 * ASTERISK-26621 - app_queue: Queue application does not ring
      members with Local interface
      (Reported by Jonas Kellens)
 * ASTERISK-26586 - chan_sip: Segfaults upon reload if client
      with MWI wasn't registered
      (Reported by Michael Kuron)
 * ASTERISK-25494 - build:  GCC 5.1.x catches some new const,
      array bounds and missing paren issues
      (Reported by George
 * ASTERISK-24499 - Need more explicit debug when PJSIP
      dialstring is invalid
      (Reported by Rusty Newton)
 * ASTERISK-25083 - Message.c: Message channel becomes saturated
      with frames leading to spammy log messages
      (Reported by
      Jonathan Rose)
 * ASTERISK-26653 - pjproject_bundled doesn't verify already
      downloaded tarballs
      (Reported by George Joseph)
 * ASTERISK-26433 - chan_sip: Allows To-tag checks to be
      bypassed, setting up new calls
      (Reported by Walter Doekes)
 * ASTERISK-26579 - codec_opus: Recursiveness when parsing fmtp
      (Reported by Jørgen H)
 * ASTERISK-26644 - PJSIPShowRegistrationsInbound just dumps all
      (Reported by George Joseph)
 * ASTERISK-26647 - Support older DNS style for OpenBSD
      (Reported by snuffy)
 * ASTERISK-26490 - res_pjsip: sends 481 Call/Transaction Does
      Not Exist when transaction branch parameter contains "_"
      (Reported by Juris Breicis)
 * ASTERISK-26629 - tests/manager: 4 test failures as a result
      of iostream change
      (Reported by Joshua Colp)
 * ASTERISK-26109 - Asterisk fails building with OpenSSL 1.1.0
      (Reported by Tzafrir Cohen)
 * ASTERISK-26617 - res_rtp_asterisk: Can't bind on systems
      without IPv6
      (Reported by Guido Falsi)
 * ASTERISK-26603 - [patch] chan_pjsip: not switching sending
      codec to receiving codec when asymmetric_rtp_codec=no
      (Reported by Alexei Gradinari)
 * ASTERISK-24330 - Requirement for 'wss' value in Contact
      header transport parameter on inbound traffic violates RFC7118
      (Reported by Marek Cervenka)
 * ASTERISK-26566 - res_rtp_asterisk: RTT miscalculation in
      (Reported by Hector Royo Concepcion)
 * ASTERISK-26604 - chan_sip: sip reload doesn't apply changes
      to tlscertfile, tlsciphers, etc.
      (Reported by Michael
 * ASTERISK-26608 - Compile and link failures on OpenBSD
      (Reported by snuffy)
 * ASTERISK-26520 - codec_opus: Generated fmtp line has no
      (Reported by Sebastian Gutierrez)
 * ASTERISK-26605 - codec_opus: Spammed warning when Opus
      negotiated but codec_opus not loaded.
      (Reported by Richard
 * ASTERISK-26516 - pjsip: Memory corruption with possible
      memory leak.
      (Reported by Richard Mudgett)
 * ASTERISK-24515 - Unconditional use of fopencookie() /
      funopen() is non-portable
      (Reported by Timo Teräs)
 * ASTERISK-26556 - manager: AMI version report same in Ast 13 &
      14, despite Ast 14 syntax changes
      (Reported by Michelle
 * ASTERISK-26592 - Latest libedit (3.1) defaults to unicode and
      makes asterisk CLI read garbage
      (Reported by George
 * ASTERISK-26575 - testsuite: Need to check PJSIP functionality
      when res_srtp is not loaded.
      (Reported by Joshua Colp)
 * ASTERISK-26565 - chan_unistim on 11, 13, 14 placing call on
      hold temporarily locks up set
      (Reported by Jason)
 * ASTERISK-26573 - Some typos in documentation of chan_sip.c
      (Reported by C.J. Collier)
 * ASTERISK-26571 - res_pjsip: Resolution incorrect when
      explicit IPv6 transport configured
      (Reported by Joshua
 * ASTERISK-26468 - ari: Bridge events stop working after this
      sequence of ARI calls
      (Reported by Daniele Pallastrelli)
 * ASTERISK-24400 - ooh323 sends wrong hangup code
      (Reported by Dmitry Melekhov)
 * ASTERISK-26555 - Multi-party Video: Fix some post Asterisk-11
      (Reported by Matt Jordan)
 * ASTERISK-26412 - build:  Prepare for gcc 6.2
      by George Joseph)
 * ASTERISK-26509 - A few non-critical deprecation warnings when
      building on Ubuntu 16.10
      (Reported by Jonathan Harris)
 * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects
      incoming calls after 2 minutes - rtptimeout behaving badly -
      (Reported by Michael Keuter)
 * ASTERISK-26549 - app_dial: When PickupChan() is used some
      channels may have incorrect device state
      (Reported by
      Joshua Colp)
 * ASTERISK-24274 - [patch]Codec Format Is Not Included in the
      SDP Media Attributes When SLIN48 Codec Is Used
      by Frankie Chin)
 * ASTERISK-26311 - [patch] rtp_engine: Allow more than 32
      dynamic payload types.
      (Reported by Alexander Traud)
 * ASTERISK-26546 - mips64el and x32 - undefined reference to
      symbol 'dlopen@@GLIBC_2.2'
      (Reported by Tzafrir Cohen)
 * ASTERISK-26541 - res_pjsip_sdp_rtp: Restrict number of
      formats to maximum
      (Reported by Joshua Colp)
 * ASTERISK-26476 - chan_sip: Incorrect display option "Outbound
      reg. retry 403" in "sip show settings"
      (Reported by Sergey
 * ASTERISK-25070 - Fix FTBFS on Hurd
      (Reported by
      Gabriele Giacone)
 * ASTERISK-26537 - AMI: NewConnectedLine event is not
      (Reported by Etienne Lessard)
 * ASTERISK-26526 - [UBSAN] vector.h: null pointer can be passed
      as argument 2 to memcpy
      (Reported by Badalian Vyacheslav)
 * ASTERISK-26524 - astobj2: data_size variable is wasted space
      when AO2_DEBUG is not enabled.
      (Reported by Corey Farrell)
 * ASTERISK-26344 - Asterisk 13.11.0 + PJSIP crash
      (Reported by Ian Gilmour)
 * ASTERISK-26387 - Asterisk segfaults shortly after starting
      even with no active calls. 
      (Reported by Harley Peters)
 * ASTERISK-26506 - [patch]res_pjsip_outbound_publish: Crash
      when publishing, in publisher_client_send at
      (Reported by Matt Krokosz)
 * ASTERISK-26513 - tests/channels/pjsip/qualify/auth: Crashing
      enough to be a nuisance
      (Reported by Joshua Colp)
 * ASTERISK-26514 - Super Awesome Company: Don't specify
      transport in pjsip.conf
      (Reported by Rusty Newton)
 * ASTERISK-26510 - pjproject_bundled uses the
      --strip-components option of tar which isn't supported in older
      (Reported by George Joseph)
 * ASTERISK-22480 - Embedded pjproject: build.mak contains
      hardcoded full path to version.mak
      (Reported by Matt
 * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes
      File not Module
      (Reported by Alexander Traud)
 * ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing
      (Reported by Bill Brigden)
 * ASTERISK-26503 - app_voicemail: Asterisk crashes when
      MailboxExists is used
      (Reported by Doug Lytle)
 * ASTERISK-26423 - res_pjsip_sdp_rtp: Asymmetric RTP codec can
      cause audio loss and wonkiness
      (Reported by Andreas
 * ASTERISK-26309 - [patch] res_pjsip: Allow IPv4/IPv6 (Dual
      Stack) installations.
      (Reported by Alexander Traud)
 * ASTERISK-26482 - [patch] chan_pjsip: segfault on already
      disconnected session
      (Reported by Alexei Gradinari)
 * ASTERISK-26455 - cdr_radius / cel_radius: try fix memory
      (Reported by Badalian Vyacheslav)
 * ASTERISK-26421 - Segmentation Fault with ARI originate into
      mixing bridge with 43 clients
      (Reported by Andrew Nagy)
 * ASTERISK-26444 - 'features show' command in CLI does not
      return prompt.
      (Reported by John Kiniston)
 * ASTERISK-26356 - menuselect: invalid test for GTK2
      (Reported by Tzafrir Cohen)
 * ASTERISK-26477 - pjproject:  SEGV during SSL operations
      (Reported by George Joseph)
 * ASTERISK-26462 - [patch] app_queue: While using queues with
      realtime, setting back to an empty context doesn't stop the exit
      key usage
      (Reported by Leandro Dardini)
 * ASTERISK-26439 - chan_rtp: Crash when originating
      (Reported by Kayode)
 * ASTERISK-17470 - [patch] - When videosupport=yes, asterisk 
      allows one end peer to send video, even though the other end
      supports only audio.
      (Reported by effie mouzeli)
 * ASTERISK-26416 - pjproject-bundled: configure fails to check
      for all required utilities
      (Reported by Corey Farrell)
 * ASTERISK-26466 - core: Be forgiving on external callerid that
      may be flawed so we don't drop events
      (Reported by Richard
 * ASTERISK-26362 - res_config_mysql:  Broken after 13.10
      (Reported by Carlos Chavez)
 * ASTERISK-26446 - app_dial:  There's no way to override the
      hangupcause on unanswered channels
      (Reported by George
 * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT
      detection triggered.
      (Reported by Alexander Traud)
 * ASTERISK-26453 - res_pjsip_config_wizard: Memory leak in
      (Reported by Badalian Vyacheslav)
 * ASTERISK-26410 - core: Asterisk 14 doesn't show the header in
      the console or verbose when starting
      (Reported by Dan
 * ASTERISK-24311 - Populating database via Alembic fails when
      using same database for multiple schema sets
      (Reported by
      Dafi Ni)
 * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT =
      No Symmetric Response.
      (Reported by Alexander Traud)
 * ASTERISK-26330 - app_queue: Changing the "ringinuse"
      parameter of a queue doesn't affect dynamic members
      (Reported by Etienne Lessard)
 * ASTERISK-26426 - format_ogg_opus: remove from source
      (Reported by Kevin Harwell)
 * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when
      (Reported by Jacek)
 * ASTERISK-25468 - Deadlock in chan_sip - core show locks shows
      do_monitor lock
      (Reported by Barry Flanagan)
 * ASTERISK-26397 - manager: PresenceState action crashes
      Asterisk 14
      (Reported by Andrew Nagy)
 * ASTERISK-26389 - res_odbc: Clean up pooling options
      (Reported by Joshua Colp)
 * ASTERISK-26273 - core: Won't compile when LOW_MEMORY is
      (Reported by Anthony Messina)
 * ASTERISK-26391 - Consoles do not display verbose logger
      messages even when requested.
      (Reported by Marcelo Terres)
 * ASTERISK-26352 - Astcanary dies when doing "core restart"
      (Reported by Walter Doekes)
 * ASTERISK-19867 - asterisk fails to lower its priority when
      astcanary dies
      (Reported by Xavier Hienne)
 * ASTERISK-26263 - SQL error when using realtime and
      registering extension / inserting into ps_contacts
      (Reported by Jeppe Ryskov Larsen)
 * ASTERISK-26365 - rtp: Offer with multiple payloads for same
      codec is incorrectly handled
      (Reported by Joshua Colp)
 * ASTERISK-26374 - res_pjsip_multihomed: Contact port is
      rewritten for connectionful protocols
      (Reported by Joshua
 * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so
      (Reported by Tzafrir Cohen)
 * ASTERISK-26367 - rtp: Timestamps broken when video frame is
      across multiple RTP packets
      (Reported by Joshua Colp)
 * ASTERISK-26375 - res_pjsip_transport_management: Log message
      states seconds, but time value is milliseconds
      by Joshua Colp)
 * ASTERISK-19968 - TCP Session-Timers not dropping call
      (Reported by Aaron Hamstra)
 * ASTERISK-26364 - res_pjsip: Don't assume a request will have
      target addresses
      (Reported by Joshua Colp)
 * ASTERISK-26360 - app_queue: "queue show" output gets "failed
      to extend from 240 to 327" msgs.
      (Reported by Richard
 * ASTERISK-26358 - chan_sip: Contact is updated on re-200, but
      not on re-INVITE
      (Reported by Walter Doekes)
 * ASTERISK-26316 - res_pjsip_callerid: Irregular URI causes
      unexpected callerid
      (Reported by Kevin Harwell)
 * ASTERISK-26349 -  13.11.1 res_pjsip/pjsip_distributor.c:
      Request 'REGISTER' failed
      (Reported by Dmitry Melekhov)
 * ASTERISK-26317 - res_pjsip_session: Add ability to use
      preferred codec only
      (Reported by Aaron An)
 * ASTERISK-26264 - res_pjsip: Crash when applying ACL from
      non-existent endpoint
      (Reported by nappsoft)
 * ASTERISK-26272 - chan_sip: File descriptors leak (UDP
      (Reported by Etienne Lessard)
 * ASTERISK-20234 - SRTP not working with some devices (Eg
      snom320) - Message "We are requesting SRTP for audio, but they
      responded without it!"
      (Reported by tootai)
 * ASTERISK-26341 - ARI: Stopping a media playlist only stops
      the current media URI being played back, and not the whole list

      (Reported by Matt Jordan)
 * ASTERISK-26291 - res_pjsip_session: segfault on already
      disconnected session
      (Reported by Alexei Gradinari)
 * ASTERISK-23989 - [patch]SDP offer/answer fails if crypto keys
      added to non-crypto offer
      (Reported by Olle Johansson)
 * ASTERISK-25691 - Crash occurs when screening mode (Dial's 'p'
      argument) is enabled and callee rejects a call or hangs up.
      (Reported by Etienne Lessard)
 * ASTERISK-26331 - Crash on “core show channeltype Surrogate”
      in ast_format_cap_get_names
      (Reported by CGI.NET)
 * ASTERISK-26085 - app_mp3: results in timeout for streams
      (Reported by Jens Bürger)
 * ASTERISK-26269 - res_pjsip: Wrong state for aors without
      registered contacts after startup
      (Reported by nappsoft)
 * ASTERISK-26226 - pbx: Asterisk crash on AMI action
      "ShowDialplan" when there's a circular dependency between
      (Reported by Etienne Lessard)
 * ASTERISK-26299 - app_queue: Queue application sometimes stops
      calling members with Local interface
      (Reported by Etienne
 * ASTERISK-26279 - pjproject-bundled:  Fails to compile on
      Debian 6
      (Reported by George Joseph)
 * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not
      cleaning up properly
      (Reported by Alexander Traud)
 * ASTERISK-26203 - res_fax: Deadlock when using
      FAXOPT(gateway)=yes with Local channels
      (Reported by
      Etienne Lessard)
 * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates
      locking inversion in T.38 query option with features bridging
      (Reported by David Brillert)
 * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with
      local channels.
      (Reported by Richard Mudgett)
 * ASTERISK-26282 - AEL: macro-call in Dial application, macro
      "lacks 's' extension"
      (Reported by chris de rock)
 * ASTERISK-22820 - [patch] Plaintext auth is still supported in
      (Reported by Eugene)
 * ASTERISK-22374 - Finish mapping the sip.conf parameters to
      res_sip.conf parameters
      (Reported by Matt Jordan)
 * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of
      SSLv3, security fix POODLE (CVE-2014-3566)
      (Reported by
 * ASTERISK-26228 - res_pjsip_sdp_rtp: G729A does not include
      annexb=no attribute.
      (Reported by Ali Ghavidel)
 * ASTERISK-25472 - Swagger scripts are not replacing format
      variable in file brief
      (Reported by Corey Farrell)
 * ASTERISK-25984 - res_odbc relies on res_odbc_transaction, but
      it's not mandatory to compile it
      (Reported by József
 * ASTERISK-26305 - Asterisk 14: Two resolver unbound testsuite
      tests fail
      (Reported by Richard Mudgett)
 * ASTERISK-26288 - followme: fails to reset config items to
      default values on reload
      (Reported by Tzafrir Cohen)
 * ASTERISK-26303 - [patch] BuildSystem: ca_list_path
      capabilities not detected in PJProject.
      (Reported by
      Alexander Traud)
 * ASTERISK-25492 - ARI: Path parameters are case sensitive
      (Reported by Joshua Colp)
 * ASTERISK-26164 - XMPP no longer triggers NOTIFY to device via
      (Reported by Ross Beer)
 * ASTERISK-26233 - pbx: Failure to remove inconsistent
      extension names
      (Reported by Corey Farrell)
 * ASTERISK-26246 - Security: Privilege escalation by AMI adding
      dialplan extensions.
      (Reported by Richard Mudgett)
 * ASTERISK-26267 - ast_register_atexit callbacks should be run
      on failed startup.
      (Reported by Corey Farrell)
 * ASTERISK-26241 - res_pjsip:  When using compact headers, rpid
      and pai are incorrectly generated
      (Reported by George
 * ASTERISK-25797 - app_queue: Crash when calling a queue with a
      member with a forward to an nonexistent extension
      (Reported by Etienne Lessard)
 * ASTERISK-26239 - res_pjsip_logger:  An empty global/debug
      option is treated as a "match all" hostname
      (Reported by
      George Joseph)
 * ASTERISK-26238 - res_pjsip: Empty global default_from_user
      causes crash
      (Reported by Joshua Colp)
 * ASTERISK-26268 - alembic: 'auth_username' not in PJSIP
      'identify_by' enum
      (Reported by Joshua Colp)
 * ASTERISK-26253 - sdp_srtp: libsrtp now a required dependency,
      shouldn't be
      (Reported by Ben Merrills)
 * ASTERISK-26145 - pjsip: Deadlock with suspend + masquerade +
      (Reported by Ross Beer)
 * ASTERISK-26183 - alembic: error when using sqlalchemy version
      (Reported by Kevin Harwell)
 * ASTERISK-26283 - res_resolver_unbound:  fails configure on
      older Ubuntu and CentOS
      (Reported by George Joseph)
 * ASTERISK-26280 - DNS lookups can block channel media paths
      (Reported by Mark Michelson)
 * ASTERISK-26278 - asterisk.h should produce a reasonable error
      for external modules that fail to define AST_MODULE_SELF_SYM.
      (Reported by Corey Farrell)
 * ASTERISK-25217 - [patch]res_pjsip_outbound_publish.c needs a
      similar treatment for module unloading as
      (Reported by Richard
 * ASTERISK-26265 - Errors ignored from some parts of system
      (Reported by Corey Farrell)
 * ASTERISK-26206 - [patch] res_pjsip: Use more compatible regex
      for get all
      (Reported by Dmitry Wagin)
 * ASTERISK-26256 - [patch] SIP/SDP origin (o=) contains
      brackets with IP6
      (Reported by Alexander Traud)
 * ASTERISK-25996 - Remove "live_dangerously" requirement on
      (Reported by Andrew Nagy)
 * ASTERISK-26148 - pjsip: Cannot compile 13.10.0-rc1:
      "libasteriskpj.so: undefined reference to..."
      (Reported by
      Hans van Eijsden)
 * ASTERISK-26237 - Fax is detected on regular calls.
      (Reported by Richard Mudgett)
 * ASTERISK-26227 - sqlalchemy error due to long identifier
      (Reported by Mark Michelson)
 * ASTERISK-14 - asterisk leaves zombie mpg123
      by dcarr)
 * ASTERISK-23013 - [patch] Deadlock between 'sip show channels'
      command and attended transfer handling
      (Reported by Ben
 * ASTERISK-26199 - PJSIP: tx_data_destroy called twice
      (Reported by Scott Griepentrog)
 * ASTERISK-26166 - res_pjsip_pubsub: Crash when decrementing
      reference count of message
      (Reported by Ross Beer)
 * ASTERISK-26174 - res_pjsip: Crash when freeing cloned message
      in distributor
      (Reported by Ross Beer)
 * ASTERISK-26216 - res_fax: Deadlock when detect fax while
      channel executing Playback
      (Reported by Richard Mudgett)
 * ASTERISK-26214 - Allow arbitrary time for fax detection to
      end on a channel
      (Reported by Richard Mudgett)

New Features made in this release:
 * ASTERISK-27063 - Add support for systemd socket activation
      (Reported by Corey Farrell)
 * ASTERISK-27117 - core: Add support for timelen parsing to
      ast_parse_arg and ACO.
      (Reported by Corey Farrell)
 * ASTERISK-27129 - ast_waitfordigit_full: add support for
      filtering DTMF keys which can break the wait.
      (Reported by
      Corey Farrell)
 * ASTERISK-26995 - Add QUEUE_FLOAT_PENALTY to app_queue
      (Reported by Steve Davies)
 * ASTERISK-26878 - func_channel: Add ability to get the callid
      so dialplan has access to it.
      (Reported by Richard
 * ASTERISK-26863 - res_pjsip: Add endpoint identification
      scheme based on a configured SIP header/value
      (Reported by
      Matt Jordan)
 * ASTERISK-17428 - [patch] Allow "Comedian Mail" branding to be
      (Reported by John Covert)
 * ASTERISK-26584 - [patch] RTCP feedback for codec modules
      (Reported by Lorenzo Miniero)
 * ASTERISK-19862 - app_queue: Update Data of Queues (use queues
      as outbound calls container)
      (Reported by Sebastian
 * ASTERISK-26630 - Make logging PJPROJECT messages a bit
      (Reported by Richard Mudgett)
 * ASTERISK-26587 - app_originate: Add option to execute gosub
      prior to dial
      (Reported by dkerr)
 * ASTERISK-26595 - ARI: Add the ability to control the source
      of video in a multi-party mixing bridge
      (Reported by Matt
 * ASTERISK-26492 - ARI: Add ability to specify channel
      variables on websocket events
      (Reported by Mark Michelson)
 * ASTERISK-26470 - ARI: Add an 'asterisk_id' field to outgoing
      (Reported by Matt Jordan)
 * ASTERISK-26277 - Add dialplan function
      PJSIP_SEND_SESSION_REFRESH that sends a session refresh to
      update formats on a channel after session establishment
      (Reported by Matt Jordan)

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

Thank you for your continued support of Asterisk!
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