[asterisk-announce] Certified Asterisk 13.18-cert1 Now Available

Asterisk Development Team asteriskteam at digium.com
Thu Dec 21 16:05:52 CST 2017

The Asterisk Development Team would like to announce the release of Certified Asterisk 13.18-cert1.
This release is available for immediate download at

The release of Certified Asterisk 13.18-cert1 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-27355 - Upgrade bundled PJPROJECT to 2.7
      (Reported by Richard Mudgett)
 * 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)
 * ASTERISK-27066 - res_pjsip: Add DTMF INFO Failback mode
      (Reported by Torrey Searle)
 * ASTERISK-26230 - [patch] res_pjsip_mwi: unsolicited mwi could
      block PJSIP taskprocessor on startup
      (Reported by Alexei
 * ASTERISK-27068 - app_voicemail: Add global option
      "imap_poll_logout" to specify post-polling disconnect
      (Reported by Alexei Gradinari)
 * 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-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-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-23828 - pjsip - Need a command to list active SIP
      (Reported by Rusty Newton)
 * ASTERISK-26527 - Testsuite: increase timeout to check "core
      fullybooted wait" up to 30 sec
      (Reported by Badalian
 * 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)

Bugs fixed in this release:
 * ASTERISK-27430 - README refers to security documents that do
      not exist.
      (Reported by Corey Farrell)
 * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
      FXS gateway
      (Reported by Tzafrir Cohen)
 * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
      RTCP packet will write past where it should
      (Reported by
      Vitezslav Novy)
 * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
      not applied on reload
      (Reported by John Bigelow)
 * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
      Variable CDR(amaflags)=...
      (Reported by Richard Mudgett)
 * ASTERISK-27421 - RTP source learning not working with devices
      that have some clock issues
      (Reported by nappsoft)
 * ASTERISK-27453 - RTP: Blind transfer direct media scenario
      results in one way audio.
      (Reported by Richard Mudgett)
 * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
      flooded with unauthenticated requests
      (Reported by George
 * ASTERISK-27454 - res_http_post: Don't require
      (Reported by Joshua Colp)
 * ASTERISK-27411 - pjsip: TCP connections may not be destroyed

      (Reported by Joshua Colp)
 * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
      (Reported by Corey Farrell)
 * ASTERISK-27337 - chan_sip: Security vulnerability with client
      code header (revisited)
      (Reported by Richard Mudgett)
 * ASTERISK-27319 - (Security) Function in PJSIP 2.7
      miscalculates the length of an unsigned long variable in 64bit
      (Reported by Kim youngsung)
 * ASTERISK-27391 - Regression: Deadlock between AOR named lock
      and pjproject grp lock
      (Reported by shaurya jain)
 * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
      character isn't allowed any more
      (Reported by Michael
 * 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-26988 - res_pjsip_session: user_eq_phone adds double
      user=phone parameters to URIs
      (Reported by dtryba)
 * ASTERISK-27270 - cdr_mysql: various crashes at second module
      reload if cdr_mysql.conf is configured
      (Reported by
      Tzafrir Cohen)
 * ASTERISK-27301 - [patch] app_queue: Music On Hold for
      real-time queues is not reset to default
      (Reported by
      Nathan Bruning)
 * 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-27296 - [patch] False positive busy checks when
      icalendar's recurrence-id mechanism is involved
      by Benoît Dereck-Tricot)
 * ASTERISK-27284 - Status of RFC 3323 and PJSIP
      by dtryba)
 * ASTERISK-27216 - app_queue: does its
      check-makeannouncement-logic twice each head-caller-loop
      (Reported by Stefan Engström)
 * 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-27165 - CDR: CDR(start,u) function won't work in
      cdr_custom config
      (Reported by Jacek Konieczny)
 * 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-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-27168 - alembic: PJSIP scripts are missing column
      dtls_fingerprint in ps_endpoints table
      (Reported by
      Florian Floimair)
 * ASTERISK-23608 - ControlPlayback fails to play files with
      names containing certain non-alpha characters
      (Reported by
      Jonathan White)
 * 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-27209 - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE
      is used
      (Reported by Torrey Searle)
 * 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-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-26745 - Asymmetric codecs when
      (Reported by Jesse Ross)
 * 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-27124 - app_playback.c:say_date_generic use
      timezonename parameter
      (Reported by Holger Hans Peter
 * ASTERISK-27133 - res_rtp_asterisk: RTCP does not use ICE when
      RTCP-MUX in use
      (Reported by Joshua Colp)
 * 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-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-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-27108 - Crash using 'data get' CLI command
      (Reported by Sean Bright)
 * 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-27097 - pjproject_bundled:  We don't pass options
      needed for cross-compile to pjproject configure
      by George Joseph)
 * ASTERISK-27095 - chan_pjsip: When connected_line_method is
      set to invite, we're not trying UPDATE
      (Reported by George
 * 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-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-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-24052 - app_voicemail reloads result in leaked IMAP
      (Reported by Louis Jocelyn Paquet)
 * ASTERISK-27051 - res_pjsip_mwi: unsolicited MWI has to be
      unsubscribed on deleting the endpoint's last contact
      (Reported by Alexei Gradinari)
 * ASTERISK-27059 - res_stasis: Stolen channel references are
      (Reported by George Joseph)
 * ASTERISK-27060 - Comment typo format_g729.c
      by Matthew Fredrickson)
 * ASTERISK-26919 - res_pjsip_dialog_info_body_generator:
      Ringing&&InUse behavior difference between chan_sip and
      (Reported by Zach R)
 * 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-27041 - Core/PBX: [patch] Deadlock between dialplan
      execution and application unregistration
      (Reported by
      Frederic LE FOLL)
 * ASTERISK-27057 - Seg Fault in ast_sorcery_object_get_id at
      (Reported by Ryan Smith)
 * ASTERISK-27046 - res_pjsip_transport_websocket: segfault in
      (Reported by Jørgen H)
 * 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-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-25101 - DTLS configuration can not be specified in
      the general section - documentation
      (Reported by Ben
 * 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-26964 - res_pjsip_session: Wrong From on reinvite
      when request and To URI differ
      (Reported by Yasin CANER)
 * ASTERISK-26938 - Heap overflow in CSEQ header parsing affects
      Asterisk chan_pjsip and PJSIP
      (Reported by Sandro Gauci)
 * 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-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-26333 - Problems with Blind Transfer, PJSIP (Aastra
      (Reported by Matthias Binder)
 * 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-26143 - res_rtp_asterisk: One way audio when
      (Reported by Henning Holtschneider)
 * 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-26860 - Upon RTCP reception, netsock2.c:210
      ast_sockaddr_split_hostport: Port missing in (null)
      (Reported by Evers Lab)
 * 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-25823 - SIGSEGV, Segmentation fault. -
      ../sysdeps/x86_64/strlen.S: No such file or directory.
      (Reported by Andreas Krüger)
 * ASTERISK-26951 - chan_sip: ACK with SDP does not update a
      direct media bridge
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-26930 - pjproject/Makefile.rules for pjsip 2.6 build
      fails for non-SSE2 instrunction Linux
      (Reported by
 * ASTERISK-26926 - func_speex: Crash caused by frame with no
      (Reported by Richard Kenner)
 * ASTERISK-26929 - pjsip: Add database tables for RLS
      (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-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-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-26528 - [UBSAN] strings.h:signed integer overflow in
      (Reported by Badalian Vyacheslav)
 * 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-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-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-24712 - xmpp: starttls problem causes connection
      (Reported by Matthias Urlichs)
 * 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-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-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-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-26841 - chan_sip: Call not cancelled after receiving
      a 422 response
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-26822 - pjsip/cli_commands: pjsip show channelstats
      shows wrong codec
      (Reported by Kevin Harwell)
 * 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-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-25628 - res_config_pgsql: should match the behavior
      of other drivers so that queue_log can disable adaptive logging

      (Reported by Dmitry Wagin)
 * 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-26313 - chan_sip : Asterisk restart seems to be
      required for changing encryption option
      (Reported by
 * 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-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-26669 - PJSIP Segfault 13.13.1 (Bundled PJSIP)
      (Reported by Nic Colledge)
 * ASTERISK-18731 - [patch] DUNDi weight parameter not processed
      (Reported by Peter Racz)
 * ASTERISK-26580 - [patch] Error during LDAP modify action when
      user unregisters
      (Reported by Nicholas John Koch)
 * ASTERISK-26799 - res_pjsip: Using an auth object for inbound
      and outbound authentication fails.
      (Reported by Richard
 * 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-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-26109 - Asterisk fails building with OpenSSL 1.1.0
      (Reported by Tzafrir Cohen)
 * ASTERISK-26723 - VoiceMailPlayMsg not playing messages via
      (Reported by Ryan Rittgarn)
 * ASTERISK-18286 - [patch] 'Silence' is truncated in Record()
      (Reported by var)
 * 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-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-26704 - res_odbc.conf contains deprecated
      configuration: 'pooling', 'shared_connections', 'limit', and
      'idlecheck' options were replaced by 'max_connections'.
      (Reported by Anthony Messina)
 * ASTERISK-21094 - MixMonitorMute mutes through stream if
      already slinear (e.g. Originate)
      (Reported by David
 * 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-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-26753 - AMI disconnect causes "ast_careful_fwrite:
      fwrite() returned error: Broken pipe"
      (Reported by Kirill
 * 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-26740 - voicemail API test: uses varlibdir instead
      of datadir for a sound file
      (Reported by Tzafrir Cohen)
 * ASTERISK-26739 - voicemail API test: confuses expected and
      actual values
      (Reported by Tzafrir Cohen)
 * ASTERISK-26731 - res_sorcery_memory_cache: memory leak on
      every sorcery memory cache populate
      (Reported by Ustinov
 * 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-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-26490 - res_pjsip: sends 481 Call/Transaction Does
      Not Exist when transaction branch parameter contains "_"
      (Reported by Juris Breicis)
 * ASTERISK-26617 - res_rtp_asterisk: Can't bind on systems
      without IPv6
      (Reported by Guido Falsi)
 * ASTERISK-24330 - Requirement for 'wss' value in Contact
      header transport parameter on inbound traffic violates RFC7118
      (Reported by Marek Cervenka)
 * ASTERISK-26546 - mips64el and x32 - undefined reference to
      symbol 'dlopen@@GLIBC_2.2'
      (Reported by Tzafrir Cohen)
 * 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-26603 - [patch] chan_pjsip: not switching sending
      codec to receiving codec when asymmetric_rtp_codec=no
      (Reported by Alexei Gradinari)
 * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects
      incoming calls after 2 minutes - rtptimeout behaving badly -
      (Reported by Michael Keuter)
 * ASTERISK-26503 - app_voicemail: Asterisk crashes when
      MailboxExists is used
      (Reported by Doug Lytle)

New Features made in this release:
 * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
      incoming INVITE Request-URI.
      (Reported by Richard Mudgett)
 * 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)
 * 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-26630 - Make logging PJPROJECT messages a bit
      (Reported by Richard Mudgett)

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

Thank you for your continued support of Asterisk!
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-announce/attachments/20171221/1770a361/attachment-0001.html>

More information about the asterisk-announce mailing list