[asterisk-announce] Asterisk 14.7.0 Now Available

Asterisk Development Team asteriskteam at digium.com
Mon Oct 30 13:47:21 CDT 2017

The Asterisk Development Team would like to announce the release of Asterisk 14.7.0.
This release is available for immediate download at

The release of Asterisk 14.7.0 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-27278 - [patch] chan_sip: Provide access to read the
      full SIP Request-URI from INVITE
      (Reported by David J.
 * ASTERISK-27255 - alembic: Add support for Microsoft SQL
      (Reported by Florian Floimair)
 * ASTERISK-27253 - [patch] libsrtp-2.1.x support
      (Reported by Alexander Traud)
 * ASTERISK-27220 - Enable CHANNEL function to get from and to
      tag from SIP Headers
      (Reported by Andre Nazario)
 * ASTERISK-27169 - Google OAuth 2.0 support for XMPP / Motif
      (Reported by Andrey)
 * ASTERISK-27173 - Support for GMIME 3.0
      (Reported by
      Tzafrir Cohen)
 * ASTERISK-27092 - [patch] app_queue: Add Priority to AMI
      (Reported by Niklas Larsson)
 * ASTERISK-27085 - [patch] chan_pjsip: Port SIPDtmfMode to
      (Reported by Torrey Searle)

Bugs fixed in this release:
 * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
      curl is loaded
      (Reported by Ronald Raikes)
 * ASTERISK-27372 - ARI: Node ARI client broken in latest
      versions of 13 and 14
      (Reported by Benjamin Keith Ford)
 * ASTERISK-27047 - res_pjsip: user=phone added to Anonymous
      caller-id when it shouldn't be.
      (Reported by dtryba)
 * ASTERISK-27270 - cdr_mysql: various crashes at second module
      reload if cdr_mysql.conf is configured
      (Reported by
      Tzafrir Cohen)
 * ASTERISK-25266 - Application Originate returns SUCCESS to
      ORIGINATE_STATUS upon failure to originate
      (Reported by
      Allen Ford)
 * ASTERISK-27192 - res_pjsip: Loss of SIP registrations causing
      unavailable endpoints
      (Reported by Richard Mudgett)
 * ASTERISK-27305 - res_ari: Memory leaks in ARI when using
      Content-Type: application/json
      (Reported by David Hajek)
 * ASTERISK-26922 - chan_sip: tcpbind uses wrong source address

      (Reported by Ksenia)
 * ASTERISK-27324 - [patch] Dual-Stack server cannot be used as
      IPv4 client via TCP/TLS
      (Reported by Alexander Traud)
 * ASTERISK-27317 - vector: multiple evaluation of elem in
      (Reported by Corey Farrell)
 * ASTERISK-27318 - res_pjsip_mwi: uninitialized value from
      (Reported by Corey Farrell)
 * ASTERISK-27284 - Status of RFC 3323 and PJSIP
      by dtryba)
 * ASTERISK-27296 - [patch] False positive busy checks when
      icalendar's recurrence-id mechanism is involved
      by Benoît Dereck-Tricot)
 * ASTERISK-27216 - app_queue: does its
      check-makeannouncement-logic twice each head-caller-loop
      (Reported by Stefan Engström)
 * ASTERISK-27298 - Problem with expires on pjsip /
      (Reported by Cyrille Demaret)
 * ASTERISK-27295 - Contact is improperly translated after
      (Reported by Sean Bright)
 * ASTERISK-27292 - Multiple RTP Stream Created Breaking RFC2833
      (SSRC Changes)
      (Reported by Ross Beer)
 * ASTERISK-27289 - A codeblock that maintains a bug,but maybe
      the codeblock will never run
      (Reported by Huangyx)
 * ASTERISK-27283 - Realtime config fail with PostgreSQL version
      before 9.1
      (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-27257 - bridge_native_rtp: half-way direct media
      when using early bridging
      (Reported by Jean Aunis -
 * ASTERISK-27279 - Crash in pubsub_on_rx_request NULL pointer -
      Possible PJSIP Vulnerability
      (Reported by Ross Beer)
 * ASTERISK-26606 - tcptls: Incorrect OpenSSL function call
      leads to misleading error report
      (Reported by Bob Ham)
 * ASTERISK-16898 - SRTP unprotect: authentication failure when
      RTP sequence number switches from 65535 -> 0
      (Reported by
      Marcello Ceschia)
 * ASTERISK-27274 - RTCP needs better packet validation to
      resist port scans.
      (Reported by Richard Mudgett)
 * ASTERISK-27252 - RTP: One way audio with direct media and
      (Reported by Richard Mudgett)
 * ASTERISK-25524 - module reload res_calendar.so does not
      reload everything in calendar.conf
      (Reported by Jesper)
 * ASTERISK-24588 - res_calendar does not process CalDAV from
      Owncloud [fix included]
      (Reported by Stefan Gofferje)
 * ASTERISK-25523 - res_calendar: Warning about invalid channel
      value (for notification) occurs even when event has no
      notification configured.
      (Reported by Jesper)
 * ASTERISK-21399 - RTP Multicast of L16 (type 10): Asterisk and
      wireshark disagree
      (Reported by Tzafrir Cohen)
 * ASTERISK-27248 - [patch]external_media_address and
      external_signaling_address don't always honor localnet
      (Reported by Walter Doekes)
 * ASTERISK-27217 - chan_sip: Asterisk crashing when
      subscription doesn't get set
      (Reported by Bryan Walters)
 * ASTERISK-24066 - res_smdi: convert to astobj2
      by Corey Farrell)
 * ASTERISK-17540 - SDP origin attribute modified when issuing
      re-INVITE because of directmedia=yes
      (Reported by saghul)
 * ASTERISK-27254 - alembic: prune_on_boot fix erroneous
      (Reported by Florian Floimair)
 * ASTERISK-27232 - When in queue on g722 with interruptions,
      music on hold can get stuck and no longer play
      by Jens T.)
 * ASTERISK-27024 - nat/external_media settings ignored in
      (Reported by Christopher van de Sande)
 * ASTERISK-26879 - PJSIP external_media_address ignored if no
      local_net options are provided
      (Reported by Matt Jordan)
 * ASTERISK-27165 - CDR: CDR(start,u) function won't work in
      cdr_custom config
      (Reported by Jacek Konieczny)
 * ASTERISK-27236 - Segfault ast_channel_name (chan=0x0) at
      channel_internal_api.c:478 during T.38 Fax Receive
      (Reported by Ross Beer)
 * ASTERISK-27225 - Crash when freeing dtls_cfg->cafile
      (Reported by Richard Kenner)
 * ASTERISK-27177 - ooh323c: misleading indentation in
      (Reported by Tzafrir Cohen)
 * ASTERISK-27241 - libc segfault upon entry into app_directory

      (Reported by David Moore)
 * 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-26994 - Confbridge: CBAnn channels intermittently
      become stuck when caller hangs up before recording name
      (Reported by James Terhune)
 * ASTERISK-20858 - app_minivm fails to clean up mkstemp files
      (Reported by Walter Doekes)
 * ASTERISK-16777 - several filename bugs in Record()
      (Reported by klaus3000)
 * ASTERISK-27209 - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE
      is used
      (Reported by Torrey Searle)
 * ASTERISK-27168 - alembic: PJSIP scripts are missing column
      dtls_fingerprint in ps_endpoints table
      (Reported by
      Florian Floimair)
 * ASTERISK-19103 - When using realtime queues, function
      QUEUE_MEMBER_LIST() will return an error if no other
      app/function has loaded the queues first. This problem does not
      exist if queues.conf is used.
      (Reported by Jim Van
 * ASTERISK-21241 - When using voicemail as announce only
      (maxmsg=0), the star dtmf to enter the voicemail is not honored

      (Reported by Eelco Brolman)
 * ASTERISK-27204 - [patch] app_queue: Wrong queue stat
      (Reported by sungtae kim)
 * ASTERISK-27207 - XMPP OAuth not working due to inverted
      (Reported by Michael Kuron)
 * ASTERISK-27174 - res_calendar_icalendar: Recurring events not
      being loaded from Google calendar using ical
      (Reported by
      Mark Thompson)
 * ASTERISK-27202 - If wget is not installed and "or" is not
      available, external components (excluding pjsip) are not
      (Reported by Seán C. McCord)
 * ASTERISK-27147 - Either asterisk or pjproject isn't re-using
      tcp connections (again)
      (Reported by George Joseph)
 * ASTERISK-27193 - IPv6 receive address in message doesn't
      include brackets
      (Reported by Scott Griepentrog)
 * ASTERISK-26745 - Asymmetric codecs when
      (Reported by Jesse Ross)
 * ASTERISK-27158 - [patch] res_rtp_asterisk: RTCP statistics
      are not available when native bridge is used
      (Reported by
      Torrey Searle)
 * ASTERISK-27110 - RTP session is not fully destroyed on
      channel hangup
      (Reported by Matt Jordan)
 * ASTERISK-27171 - Asterisk 15.0.0-Beta1 does not compile
      (Reported by Ira Emus)
 * ASTERISK-26659 - res_pjsip: PJSIP presence - missing braces
      around the status element in XML
      (Reported by Abraham
 * ASTERISK-27156 - Asterisk won't compile on Fedora 26 with
      devmode enabled.
      (Reported by Corey Farrell)
 * ASTERISK-27130 - Applications ARI: Unsubscribe action for
      deviceStates does not remove old subscriptions properly
      (Reported by Sergej Kasumovic)
 * 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-27142 - sounds: Conflict between files in
      asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
      (Reported by Corey Farrell)
 * 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-27128 - [patch]res_stasis_snoop: When recording a
      snoop channel (using ARI) where no media is being received, no
      recording happens when theres no media
      (Reported by Dan
 * ASTERISK-27124 - app_playback.c:say_date_generic use
      timezonename parameter
      (Reported by Holger Hans Peter
 * ASTERISK-27127 - configs: Erroneous load directive in sample
      configuration results in "Error loading module
      (Reported by HZMI8gkCvPpom0tM)
 * ASTERISK-27105 - [patch]core: when setting 'maxfiles' in
      asterisk.conf, a message is printed, even in rasterisk -x
      (Reported by Tzafrir Cohen)
 * 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-27093 - ODBC deadlocks when app_directory tries to
      play back non-existent voicemail greeting
      (Reported by
      James Terhune)

New Features made in this release:
 * ASTERISK-27215 - [patch]AMI : Add CancelAtxfer Action
      (Reported by Thomas Sevestre)
 * ASTERISK-27117 - core: Add support for timelen parsing to
      ast_parse_arg and ACO.
      (Reported by Corey Farrell)

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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