[asterisk-users] Asterisk 15.0.0-beta1 Now Available

Asterisk Development Team asteriskteam at digium.com
Wed Aug 2 11:20:19 CDT 2017

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

The release of Asterisk 15.0.0-beta1 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 beta:

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 scgm11)
 * 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 scgm11)
 * ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry
      (Reported by scgm11)
 * 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-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 scgm11)
 * 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 scgm11)
 * 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 scgm11)
 * 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 scgm11)
 * 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 scgm11)
 * 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 beta, please see the ChangeLog:

Thank you for your continued support of Asterisk!
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