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Revision 1.31 / (download) - annotate - [select for diffs], Wed Apr 3 00:32:28 2019 UTC (7 weeks, 1 day ago) by ryoon
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Revision 1.25 / (download) - annotate - [select for diffs], Wed Aug 22 09:43:03 2018 UTC (9 months ago) by wiz
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update Asterisk to 14.7.5 -- this is a bug fix and security update,
it fixes AST-2017-005, AST-2017-006, AST-2017-006, AST-2017-008,
AST-2017-009, AST-2017-010, AST-2017-011, AST-2017-012, AST-2017-013,
and AST-2017-014.  Note that several of these are related to PJSIP
which pkgsrc doesn't use.

----- 14.7.5 -----

The Asterisk Development Team would like to announce security
releases for Asterisk 13, 14 and 15, and Certified Asterisk 13.18.
The available releases are released as versions 13.18.5, 14.7.5,
15.1.5 and 13.18-cert2.

The following security vulnerabilities were resolved in these versions:

* AST-2017-014: Crash in PJSIP resource when missing a contact header
  A select set of SIP messages create a dialog in Asterisk. Those SIP messages
  must contain a contact header. For those messages, if the header was not
  present and using the PJSIP channel driver, it would cause Asterisk to crash.
  The severity of this vulnerability is somewhat mitigated if authentication is
  enabled. If authentication is enabled a user would have to first be authorized
  before reaching the crash point.

For a full list of changes in the current releases, please see the ChangeLogs:

https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5

The security advisory is available at:

https://downloads.asterisk.org/pub/security/AST-2017-014.pdf

Thank you for your continued support of Asterisk!

----- 14.7.4 -----

The Asterisk Development Team has announced security releases for
Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
security releases are released as versions 13.13-cert9, 13.18.4,
14.7.4 and 15.1.4.

The release of these versions resolves the following security
vulnerabilities:

* AST-2017-012: Remote Crash Vulnerability in RTCP Stack
  If a compound RTCP packet is received containing more than
  one report (for example a Receiver Report and a Sender
  Report) the RTCP stack will incorrectly store report
  information outside of allocated memory potentially causing
  a crash.

For a full list of changes in the current releases, please see the
ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4

The security advisories are available at:
http://downloads.asterisk.org/pub/security/AST-2017-012.html
http://downloads.asterisk.org/pub/security/AST-2017-012.pdf

Thank you for your continued support of Asterisk!

----- 14.7.3 -----

The Asterisk Development Team has announced security releases for
Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
security releases are released as versions 13.13-cert8, 13.18.3,
14.7.3 and 15.1.3.

The release of these versions resolves the following security
vulnerabilities:

* AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
  If the chan_skinny (AKA SCCP protocol) channel driver is
  flooded with certain requests it can cause the asterisk
  process to use excessive amounts of virtual memory
  eventually causing asterisk to stop processing requests of
  any kind.

For a full list of changes in the current releases, please see the
ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3

The security advisories are available at:
http://downloads.asterisk.org/pub/security/AST-2017-013.pdf

Thank you for your continued support of Asterisk!

----- 14.7.2 -----

The Asterisk Development Team would like to announce the release
of Asterisk 14.7.2.

The release of Asterisk 14.7.2 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:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
      character isn't allowed any more
      (Reported by Michael Maier)
 * ASTERISK-27391 - Regression: Deadlock between AOR named lock
      and pjproject grp lock
      (Reported by shaurya jain)

For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.7.2

Thank you for your continued support of Asterisk!

----- 14.7.0 -----

The Asterisk Development Team would like to announce the release
of Asterisk 14.7.0.

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.  Pryke)
 * ASTERISK-27255 - alembic: Add support for Microsoft SQL
      server
      (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
      QueueStatus
      (Reported by Niklas Larsson)
 * ASTERISK-27085 - [patch] chan_pjsip: Port SIPDtmfMode to
      chan_pjsip
      (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
      AST_VECTOR_ADD_SORTED.
      (Reported by Corey Farrell)
 * ASTERISK-27318 - res_pjsip_mwi: uninitialized value from
      ast_strings_match
      (Reported by Corey Farrell)
 * ASTERISK-27284 - Status of RFC 3323 and PJSIP
      (Reported by dtryba)
 * ASTERISK-27296 - [patch] False positive busy checks when
      icalendar's recurrence-id mechanism is involved
      (Reported 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 /
      outbound-publish
      (Reported by Cyrille Demaret)
 * ASTERISK-27295 - Contact is improperly translated after
      d178f497
      (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 - Prescom)
 * 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
      strictrtp=yes.
      (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
      (Reported 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
      (Reported by Jens T.)
 * ASTERISK-27024 - nat/external_media settings ignored in 14.4.1
      (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
      addons/ooh323c/src/ooSocket.c
      (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
      possible.
      (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()
      application
      (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 Meggelen)
 * 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
      calculation
      (Reported by sungtae kim)
 * ASTERISK-27207 - XMPP OAuth not working due to inverted
      logic
      (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
      installed
      (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
      asymmetric_rtp_codec=no
      (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 Liebsch)
 * 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.
      (Reported 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
      filesystem
      (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 Newton)
 * 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 Jenkins)
 * ASTERISK-27124 - app_playback.c:say_date_generic use
      timezonename parameter
      (Reported by Holger Hans Peter Freyther)
 * ASTERISK-27127 - configs: Erroneous load directive in sample
      configuration results in "Error loading module
      'res_pjsip_multihomed.so'"
      (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:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.7.0

Thank you for your continued support of Asterisk!

----- 14.6.0 -----

The Asterisk Development Team would like to announce the release
of Asterisk 14.6.0.

The release of Asterisk 14.6.0 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:

Bugs fixed in this release:
-----------------------------------
 * 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-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-25665 - Duplicate logging in queue log for EXITEMPTY events
      (Reported by Ove Aursand)
 * ASTERISK-27065 - call hangup after leaving app_queue
      (Reported by Marek Cervenka)
 * ASTERISK-26978 - rtp: Crash in ast_rtp_codecs_payload_code()
      (Reported by Ross Beer)
 * ASTERISK-24052 - app_voicemail reloads result in leaked IMAP sockets.
      (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
      (Reported by Matthew Fredrickson)
 * ASTERISK-27041 - Core/PBX: [patch] Deadlock between dialplan
      execution and application unregistration
      (Reported by Frederic LE FOLL)
 * ASTERISK-27026 - res_ari: Crash when no ari.conf
      configuration file exists
      (Reported by Ronald Raikes)
 * ASTERISK-27057 - Seg Fault in ast_sorcery_object_get_id at
      sorcery.c
      (Reported by Ryan Smith)
 * ASTERISK-27024 - nat/external_media settings ignored in
      14.4.1
      (Reported by Christopher van de Sande)
 * ASTERISK-27046 - res_pjsip_transport_websocket: segfault in
      get_write_timeout
      (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
      network
      (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
      snooping
      (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 Pitrou)
 * 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 Harris)
 * ASTERISK-26399 - app_queue: Agent not called when caller is
      parked
      (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
      negotiable
      (Reported by Stefan Engström)
 * ASTERISK-26964 - res_pjsip_session: Wrong From on reinvite
      when request and To URI differ
      (Reported by Yasin CANER)
 * ASTERISK-26789 - Audit manipulation of channel flags without
      locks
      (Reported by Joshua Colp)
 * ASTERISK-26333 - Problems with Blind Transfer, PJSIP (Aastra
      6869i)
      (Reported by Matthias Binder)

Improvements made in this release:
-----------------------------------
 * ASTERISK-26230 - [patch] res_pjsip_mwi: unsolicited mwi could
      block PJSIP taskprocessor on startup
      (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 Falsi)
 * ASTERISK-26419 - audiohooks: Remove redundant codec
      translations when using audiohooks
      (Reported by Michael Walton)
 * ASTERISK-26976 - libsrtp-2.x.x support
      (Reported by Alex)
 * ASTERISK-26124 - res_agi: Set audio format for EAGI audio
      stream
      (Reported by John Fawcett)

For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.6.0

Thank you for your continued support of Asterisk!

Revision 1.17 / (download) - annotate - [select for diffs], Mon Jan 1 21:18:17 2018 UTC (16 months, 2 weeks ago) by adam
Branch: MAIN
Changes since 1.16: +2 -2 lines
Diff to previous 1.16 (colored)

Revbump after boost update

Revision 1.16 / (download) - annotate - [select for diffs], Thu Nov 30 16:45:16 2017 UTC (17 months, 3 weeks ago) by adam
Branch: MAIN
CVS Tags: pkgsrc-2017Q4-base, pkgsrc-2017Q4
Changes since 1.15: +2 -2 lines
Diff to previous 1.15 (colored)

Revbump after textproc/icu update

Revision 1.15 / (download) - annotate - [select for diffs], Mon Sep 18 09:53:12 2017 UTC (20 months ago) by maya
Branch: MAIN
CVS Tags: pkgsrc-2017Q3-base, pkgsrc-2017Q3, pkgsrc-
Changes since 1.14: +2 -2 lines
Diff to previous 1.14 (colored)

revbump for requiring ICU 59.x

Revision 1.14 / (download) - annotate - [select for diffs], Thu Aug 24 20:03:07 2017 UTC (20 months, 4 weeks ago) by adam
Branch: MAIN
Changes since 1.13: +2 -1 lines
Diff to previous 1.13 (colored)

Revbump for boost update

Revision 1.13 / (download) - annotate - [select for diffs], Wed Jun 21 13:33:47 2017 UTC (23 months ago) by jnemeth
Branch: MAIN
CVS Tags: pkgsrc-2017Q2-base, pkgsrc-2017Q2
Changes since 1.12: +30 -27 lines
Diff to previous 1.12 (colored)

Update to Asterisk 14.5.0: this is mostly a bug fix releases with
patches for a number of security issues, several of which do not
apply to this package because they relate to PJSIP:  AST-2016-009,
AST-2016-010, AST-2017-001, AST-2017-002, AST-2017-003, and
AST-2017-004.

----- 14.5.0

The Asterisk Development Team would like to announce the release
of Asterisk 14.5.0.

The release of Asterisk 14.5.0 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:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26982 - chan_sip: rtcp_mux setting may cause ice
      completion failure/delay if client offers rtcp-mux as
      negotiable
      (Reported by Stefan Engström)
 * ASTERISK-26979 - res_rtp_asterisk: SRTP unprotect failed with
      authentication failure 10 or 110
      (Reported by Javier Riveros)
 * ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY
      events
      (Reported by Ove Aursand)
 * 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
      transcoding
      (Reported by Henning Holtschneider)
 * 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 Changes
      (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-26860 - Upon RTCP reception, netsock2.c:210
      ast_sockaddr_split_hostport: Port missing in (null)
      (Reported by Evers Lab)
 * ASTERISK-26922 - chan_sip: tcpbind uses wrong source address
      (Reported by Ksenia)
 * 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 Mudgett)
 * 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 datalen
      (Reported by Richard Kenner)
 * 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 abelbeck)
 * ASTERISK-26929 - pjsip: Add database tables for RLS
      (Reported by Joshua Colp)
 * ASTERISK-26953 - Asterisk crash if hep.conf have some missing parameters
      (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 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
      MixMonitor
      (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
      ast_str_case_hash
      (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 support
      (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:
      headers
      (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
      incorrectly
      (Reported by Joerg Sonnenberger)
 * ASTERISK-26086 - res_musiconhold: format option is not
      documented adequately
      (Reported by Jens Bürger)
 * ASTERISK-23996 - No core dumps because of res_musiconhold chdir.
      (Reported by Walter Doekes)
 * ASTERISK-24712 - xmpp: starttls problem causes connection spew
      (Reported by Matthias Urlichs)
 * ASTERISK-26814 - pjproject_bundled build fails to download
      pjproject source when using cURL
      (Reported by Gergely Dömsödi)
 * 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
 * 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-26875 - app_mixmonitor: Recording out of sync when
      183 but no RTP
      (Reported by Aaron An)

Improvements made in this release:
-----------------------------------
 * ASTERISK-26088 - Investigate heavy memory utilization by
      res_pjsip_pubsub
      (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))

For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.5.0

Thank you for your continued support of Asterisk!

----- 14.4.0

The Asterisk Development Team would like to announce the release of
Asterisk 14.4.0.

The release of Asterisk 14.4.0 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:

*New Features made in this release:*
-----------------------------------
- [ASTERISK-26878 <https://issues.asterisk.org/jira/browse/ASTERISK-26878>]
- func_channel: Add ability to get the callid so dialplan has access to it.
(Reported by Richard Mudgett)
- [ASTERISK-26863 <https://issues.asterisk.org/jira/browse/ASTERISK-26863>]
- res_pjsip: Add endpoint identification scheme based on a configured SIP
header/value
(Reported by Matt Jordan)
- [ASTERISK-17428 <https://issues.asterisk.org/jira/browse/ASTERISK-17428>]
- [patch] Allow "Comedian Mail" branding to be removed
(Reported by John Covert)

*Bugs fixed in this release:*
-----------------------------------
- [ASTERISK-26851 <https://issues.asterisk.org/jira/browse/ASTERISK-26851>]
- res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
(Reported by Richard Begg)
- [ASTERISK-26897 <https://issues.asterisk.org/jira/browse/ASTERISK-26897>]
- chan_sip: Security vulnerability with client code header
(Reported by Alex Villac Lasso)
- [ASTERISK-26916 <https://issues.asterisk.org/jira/browse/ASTERISK-26916>]
- res_pjsip: Excessive refcount reached on transport ao2 object
(Reported by Ross Beer)
- [ASTERISK-26705 <https://issues.asterisk.org/jira/browse/ASTERISK-26705>]
- libasteriskssl.so not found when asterisk is installed for the 1st time
(Reported by George Joseph)
- [ASTERISK-26850 <https://issues.asterisk.org/jira/browse/ASTERISK-26850>]
- res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field
in HEP packets
(Reported by Max Norba)
- [ASTERISK-26484 <https://issues.asterisk.org/jira/browse/ASTERISK-26484>]
- res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from'
argument.
(Reported by Vinod Dharashive)
- [ASTERISK-26776 <https://issues.asterisk.org/jira/browse/ASTERISK-26776>]
- res_pjsip_pubsub: Crash when generating xpidf content
(Reported by Andrew Green)
- [ASTERISK-26880 <https://issues.asterisk.org/jira/browse/ASTERISK-26880>]
- Asterisk crashes when multiple speex users join confbridge with pp_vad
and dtx enabled
(Reported by Kirsty Tyerman)
- [ASTERISK-26862 <https://issues.asterisk.org/jira/browse/ASTERISK-26862>]
- app_queue: Queue stops calling members with local interface after
forwarding in previous call
(Reported by Robert Mordec)
- [ASTERISK-26732 <https://issues.asterisk.org/jira/browse/ASTERISK-26732>]
- res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
(Reported by Dan Jenkins)
- [ASTERISK-26879 <https://issues.asterisk.org/jira/browse/ASTERISK-26879>]
- PJSIP external_media_address ignored if no local_net options are provided
(Reported by Matt Jordan)
- [ASTERISK-26867 <https://issues.asterisk.org/jira/browse/ASTERISK-26867>]
- autochan: Locking in a function ast_autochan_destroy() on destroyed
channel (after masquerade).
(Reported by Krzysztof Trempala)
- [ASTERISK-26869 <https://issues.asterisk.org/jira/browse/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 <https://issues.asterisk.org/jira/browse/ASTERISK-26668>]
- core: Malformed pattern matching extension (various factors) results in
crash
(Reported by xrobau)
- [ASTERISK-26865 <https://issues.asterisk.org/jira/browse/ASTERISK-26865>]
- chan_iax2: Reload of iax peer results in loss of host address/port
(Reported by Richard Begg)
- [ASTERISK-26872 <https://issues.asterisk.org/jira/browse/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)
(Reported by Matt Jordan)
- [ASTERISK-26717 <https://issues.asterisk.org/jira/browse/ASTERISK-26717>]
- Document the fact that Asterisk HEP support only works with the PJSIP
channel driver
(Reported by Olivier Krief)
- [ASTERISK-26643 <https://issues.asterisk.org/jira/browse/ASTERISK-26643>]
- Extra new line in Device field of DeviceStateChange AMI Event after
restart of Asterisk
(Reported by Roman Bedros)
- [ASTERISK-25237 <https://issues.asterisk.org/jira/browse/ASTERISK-25237>]
- stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
(Reported by Smirnov Aleksey)
- [ASTERISK-26857 <https://issues.asterisk.org/jira/browse/ASTERISK-26857>]
- chan_pjsip: Dialplan function race condition
(Reported by Joshua Colp)
- [ASTERISK-26841 <https://issues.asterisk.org/jira/browse/ASTERISK-26841>]
- chan_sip: Call not cancelled after receiving a 422 response
(Reported by Jean Aunis - Prescom)
- [ASTERISK-26822 <https://issues.asterisk.org/jira/browse/ASTERISK-26822>]
- pjsip/cli_commands: pjsip show channelstats shows wrong codec
(Reported by Kevin Harwell)
- [ASTERISK-26353 <https://issues.asterisk.org/jira/browse/ASTERISK-26353>]
- res_musiconhold: musiconhold seems to think that the general section is a
class and issues warning
(Reported by Jonathan Harris)
- [ASTERISK-26685 <https://issues.asterisk.org/jira/browse/ASTERISK-26685>]
- res_pjsip: Crash when using IPv6 and Transport ws,wss
(Reported by Michael Balen)
- [ASTERISK-24562 <https://issues.asterisk.org/jira/browse/ASTERISK-24562>]
- app_voicemail: Cannot set fromstring on a per-mailbox basis
(Reported by Mark Scholten)
- [ASTERISK-26598 <https://issues.asterisk.org/jira/browse/ASTERISK-26598>]
- Saynumber is trying to get "and" from "digits/" subfolder
(Reported by Jonathan Harris)
- [ASTERISK-17067 <https://issues.asterisk.org/jira/browse/ASTERISK-17067>]
- Long lines in call files cause spurious syntax error
(Reported by Dave Olszewski)
- [ASTERISK-26796 <https://issues.asterisk.org/jira/browse/ASTERISK-26796>]
- res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
(Reported by Jgen H)
- [ASTERISK-25628 <https://issues.asterisk.org/jira/browse/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 <https://issues.asterisk.org/jira/browse/ASTERISK-26774>]
- core: Playback URL fails after some time
(Reported by Igor Gamayunov)
- [ASTERISK-26825 <https://issues.asterisk.org/jira/browse/ASTERISK-26825>]
- pjsip.conf.sample: user_agent: still refers to branch 12
(Reported by Tzafrir Cohen)
- [ASTERISK-26823 <https://issues.asterisk.org/jira/browse/ASTERISK-26823>]
- PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not
exist
(Reported by Mark Michelson)
- [ASTERISK-26623 <https://issues.asterisk.org/jira/browse/ASTERISK-26623>]
- res_pjsip: Crash when calling PJSIPShowEndpoint
(Reported by Jgen H)
- [ASTERISK-26808 <https://issues.asterisk.org/jira/browse/ASTERISK-26808>]
- res_pjsip_outbound_registration doesn't know about network change events
(Reported by George Joseph)
- [ASTERISK-26781 <https://issues.asterisk.org/jira/browse/ASTERISK-26781>]
- bridge: Passing the 'p' (play tone) flag to Bridge() application results
in garbled audio
(Reported by Sean Bright)
- [ASTERISK-26782 <https://issues.asterisk.org/jira/browse/ASTERISK-26782>]
- res_pjsip: URI requirement for fields is not consistently documented and
error does not provide indication
(Reported by Peter Sokolov)
- [ASTERISK-26812 <https://issues.asterisk.org/jira/browse/ASTERISK-26812>]
- [patch] Fix download_externals To Allow The Use Of curl Or wget
(Reported by Michael L. Young)
- [ASTERISK-18271 <https://issues.asterisk.org/jira/browse/ASTERISK-18271>]
- Pattern matching with res_config_mysql extensions does not behave as
expected
(Reported by Charlie Smurthwaite)
- [ASTERISK-26669 <https://issues.asterisk.org/jira/browse/ASTERISK-26669>]
- PJSIP Segfault 13.13.1 (Bundled PJSIP)
(Reported by Nic Colledge)
- [ASTERISK-18731 <https://issues.asterisk.org/jira/browse/ASTERISK-18731>]
- [patch] DUNDi weight parameter not processed correctly
(Reported by Peter Racz)
- [ASTERISK-26799 <https://issues.asterisk.org/jira/browse/ASTERISK-26799>]
- res_pjsip: Using an auth object for inbound and outbound authentication
fails.
(Reported by Richard Mudgett)
- [ASTERISK-26738 <https://issues.asterisk.org/jira/browse/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
(Reported by Michael Maier)
- [ASTERISK-25893 <https://issues.asterisk.org/jira/browse/ASTERISK-25893>]
- Function vmauthenticate accesses uninitialized memory
(Reported by Filip Jenicek)
- [ASTERISK-26580 <https://issues.asterisk.org/jira/browse/ASTERISK-26580>]
- [patch] Error during LDAP modify action when user unregisters
(Reported by Nicholas John Koch)
- [ASTERISK-26802 <https://issues.asterisk.org/jira/browse/ASTERISK-26802>]
- [patch] Integrity Check Of PJSIP Download Fails
(Reported by Michael L. Young)
- [ASTERISK-15858 <https://issues.asterisk.org/jira/browse/ASTERISK-15858>]
- [patch] Fix query with double backslash in string literals and stop log
warnings
(Reported by Humberto Figuera)
- [ASTERISK-26057 <https://issues.asterisk.org/jira/browse/ASTERISK-26057>]
- res_config_sqlite3 uses incorrect query - unnecessary escape
(Reported by Stepan)
- [ASTERISK-23457 <https://issues.asterisk.org/jira/browse/ASTERISK-23457>]
- SQlite3: Realtime queue loading fails after PRAGMA query result
(Reported by Scott Griepentrog)
- [ASTERISK-26794 <https://issues.asterisk.org/jira/browse/ASTERISK-26794>]
- http: Crash on Reload Only in ast_tcptls_server_start
(Reported by Joshua Elson)
- [ASTERISK-26714 <https://issues.asterisk.org/jira/browse/ASTERISK-26714>]
- Phone default have not ringing on ARM
(Reported by Igor Goncharovsky)
- [ASTERISK-26696 <https://issues.asterisk.org/jira/browse/ASTERISK-26696>]
- pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on
subscription refresh
(Reported by Zach R)
- [ASTERISK-26756 <https://issues.asterisk.org/jira/browse/ASTERISK-26756>]
- res_pjsip_mwi: Asterisk does not terminate MWI subscription
(Reported by Carl Fortin)
- [ASTERISK-26109 <https://issues.asterisk.org/jira/browse/ASTERISK-26109>]
- Asterisk fails building with OpenSSL 1.1.0
(Reported by Tzafrir Cohen)
- [ASTERISK-26723 <https://issues.asterisk.org/jira/browse/ASTERISK-26723>]
- VoiceMailPlayMsg not playing messages via realtime
(Reported by Ryan Rittgarn)
- [ASTERISK-18286 <https://issues.asterisk.org/jira/browse/ASTERISK-18286>]
- [patch] 'Silence' is truncated in Record()
(Reported by var)
- [ASTERISK-26248 <https://issues.asterisk.org/jira/browse/ASTERISK-26248>]
- chan_pjsip: Error when calling PJSIP client with domain specified
(Reported by Norbert Varga)
- [ASTERISK-26788 <https://issues.asterisk.org/jira/browse/ASTERISK-26788>]
- core: Protect flags during ast_waitfor
(Reported by Joshua Colp)
- [ASTERISK-26115 <https://issues.asterisk.org/jira/browse/ASTERISK-26115>]
- pbx: AMI Originate ignore "failed" extension on call failure
(Reported by Nasir Iqbal)
- [ASTERISK-26785 <https://issues.asterisk.org/jira/browse/ASTERISK-26785>]
- configs/samples: The 'identify' entry is in the wrong section in
sorcery.conf.sample
(Reported by Torrey Searle)
- [ASTERISK-26772 <https://issues.asterisk.org/jira/browse/ASTERISK-26772>]
- Crash in srv.c on startup with pjsip
(Reported by nappsoft)
- [ASTERISK-26770 <https://issues.asterisk.org/jira/browse/ASTERISK-26770>]
- res_stasis_device_state: Duplicate subscriptions when multiple received
at same time
(Reported by Joshua Colp)

*Improvements made in this release:*
-----------------------------------
- [ASTERISK-26864 <https://issues.asterisk.org/jira/browse/ASTERISK-26864>]
- res_pjsip_session: Add support for overlap dialling
(Reported by Richard Begg)
- [ASTERISK-26846 <https://issues.asterisk.org/jira/browse/ASTERISK-26846>]
- chan_sip: Add rtcp-mux support
(Reported by Sean Bright)

*Thank you for your continued support of Asterisk!*

----- 14.3.0

The Asterisk Development Team has announced the release of Asterisk 14.3.0.

The release of Asterisk 14.3.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

New Features made in this release:
-----------------------------------
 * ASTERISK-26630 - Make logging PJPROJECT messages a bit easier
      (Reported by Richard Mudgett)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26772 - Crash in srv.c on startup with pjsip (Reported
      by nappsoft)
 * 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 Woolley)
 * 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-24858 - [patch]Asterisk 13 PJSIP sends RTP packets in
      wrong byte order on Intel platform when using slin codec
      (Reported by Frankie Chin)
 * 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
      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 (Reported 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 Artem)
 * ASTERISK-26710 - [patch] res_rtp_asterisk: CHANNEL arguments,
      (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
      (Reported by Aaron An)
 * ASTERISK-26670 - [patch] Outgoing SIP-URI Dialing via PJSIP
      (Reported by Alexander Traud)
 * ASTERISK-26691 - Remember SDP negotiation on SIP_CODEC_INBOUND.
      (Reported by Alexander Traud)
 * ASTERISK-26673 - chan_pjsip: Crash when using CHANNEL dialplan
      function around masquerade (Reported by Joshua Colp)
 * ASTERISK-26684 - res_pjsip: Various issues with compact SIP
      headers (Reported by Joshua Elson)
 * ASTERISK-26655 - [patch]pjsip: Transfers Broken with Compact
      Headers Enabled (Reported by JoshE)
 * ASTERISK-26672 - Crash when setting remote address on RTP
      instance (Reported by Richard Mudgett)
 * 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 Joseph)
 * 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
      line (Reported by Jørgen H)
 * ASTERISK-26644 - PJSIPShowRegistrationsInbound just dumps all
      aors (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-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-26546 - mips64el and x32 - undefined reference to
      symbol 'dlopen@@GLIBC_2.2' (Reported by Tzafrir Cohen)
 * ASTERISK-26566 - res_rtp_asterisk: RTT miscalculation in RTCP
      (Reported by Hector Royo Concepcion)
 * ASTERISK-26604 - chan_sip: sip reload doesn't apply changes to
      tlscertfile, tlsciphers, etc. (Reported by Michael Kuron)
 * ASTERISK-26608 - Compile and link failures on OpenBSD (Reported
      by snuffy)

Improvements made in this release:
-----------------------------------
 * ASTERISK-23828 - pjsip - Need a command to list active SIP
      subscriptions (Reported by Rusty Newton)
 * ASTERISK-26527 - Testsuite: increase timeout to check "core
      fullybooted wait" up to 30 sec (Reported by Badalian Vyacheslav)
 * 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)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.3.0

Thank you for your continued support of Asterisk!

Revision 1.12 / (download) - annotate - [select for diffs], Sun Apr 30 01:21:30 2017 UTC (2 years ago) by ryoon
Branch: MAIN
Changes since 1.11: +2 -2 lines
Diff to previous 1.11 (colored)

Recursive revbump from boost update

Revision 1.11 / (download) - annotate - [select for diffs], Sat Apr 22 21:03:26 2017 UTC (2 years, 1 month ago) by adam
Branch: MAIN
Changes since 1.10: +2 -2 lines
Diff to previous 1.10 (colored)

Revbump after icu update

Revision 1.10 / (download) - annotate - [select for diffs], Sun Feb 12 06:25:09 2017 UTC (2 years, 3 months ago) by ryoon
Branch: MAIN
CVS Tags: pkgsrc-2017Q1-base, pkgsrc-2017Q1
Changes since 1.9: +2 -2 lines
Diff to previous 1.9 (colored)

Recursive revbump from fonts/harfbuzz

Revision 1.9 / (download) - annotate - [select for diffs], Mon Feb 6 13:55:09 2017 UTC (2 years, 3 months ago) by wiz
Branch: MAIN
Changes since 1.8: +2 -2 lines
Diff to previous 1.8 (colored)

Recursive bump for harfbuzz's new graphite2 dependency.

Revision 1.8 / (download) - annotate - [select for diffs], Thu Jan 19 18:52:04 2017 UTC (2 years, 4 months ago) by agc
Branch: MAIN
Changes since 1.7: +4 -4 lines
Diff to previous 1.7 (colored)

Convert all occurrences (353 by my count) of

	MASTER_SITES= 	site1 \
			site2

style continuation lines to be simple repeated

	MASTER_SITES+= site1
	MASTER_SITES+= site2

lines. As previewed on tech-pkg. With thanks to rillig for fixing pkglint
accordingly.

Revision 1.7 / (download) - annotate - [select for diffs], Sun Jan 1 16:06:06 2017 UTC (2 years, 4 months ago) by adam
Branch: MAIN
Changes since 1.6: +2 -2 lines
Diff to previous 1.6 (colored)

Revbump after boost update

Revision 1.6 / (download) - annotate - [select for diffs], Sun Dec 4 05:17:19 2016 UTC (2 years, 5 months ago) by ryoon
Branch: MAIN
CVS Tags: pkgsrc-2016Q4-base, pkgsrc-2016Q4
Changes since 1.5: +2 -1 lines
Diff to previous 1.5 (colored)

Recursive revbump from textproc/icu 58.1

Revision 1.5 / (download) - annotate - [select for diffs], Sun Nov 27 22:55:51 2016 UTC (2 years, 5 months ago) by jnemeth
Branch: MAIN
Changes since 1.4: +7 -2 lines
Diff to previous 1.4 (colored)

Update to Asterisk 14.2.0: this is mostly a bugfix release with some minor
improvements.

pkgsrc change: adapt to new res_resolver_unbound module.

The Asterisk Development Team has announced the release of Asterisk 14.2.0.

The release of Asterisk 14.2.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Improvements made in this release:
-----------------------------------
 * 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-26538 - codec_opus: Add sample to
      configs/samples/codecs.conf.sample (Reported by Kevin Harwell)
 * 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)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26608 - Compile and link failures on OpenBSD (Reported
      by snuffy)
 * ASTERISK-26520 - codec_opus: Generated fmtp line has no content
      (Reported by scgm11)
 * ASTERISK-26605 - codec_opus: Spammed warning when Opus
      negotiated but codec_opus not loaded. (Reported by Richard
      Mudgett)
 * ASTERISK-26516 - pjsip: Memory corruption with possible memory
      leak. (Reported by Richard Mudgett)
 * ASTERISK-26556 - manager: AMI version report same in Ast 13 &
      14, despite Ast 14 syntax changes (Reported by Michelle Dupuis)
 * ASTERISK-26343 - ASTERISK-25951 causes issues for callerid
      manipulation through agi (Reported by Morten Tryfoss)
 * ASTERISK-26592 - Latest libedit (3.1) defaults to unicode and
      makes asterisk CLI read garbage (Reported by George Joseph)
 * ASTERISK-26565 - chan_unistim on 11, 13, 14 placing call on hold
      temporarily locks up set (Reported by Jason)
 * ASTERISK-26575 - testsuite: Need to check PJSIP functionality
      when res_srtp is not loaded. (Reported by Joshua Colp)
 * ASTERISK-26571 - res_pjsip: Resolution incorrect when explicit
      IPv6 transport configured (Reported by Joshua Colp)
 * 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
      regressions (Reported by Matt Jordan)
 * ASTERISK-26412 - build:  Prepare for gcc 6.2 (Reported 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 - regression
      (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 (Reported by Frankie
      Chin)
 * ASTERISK-26311 - [patch] rtp_engine: Allow more than 32 dynamic
      payload types. (Reported by Alexander Traud)
 * ASTERISK-26506 - [patch]res_pjsip_outbound_publish: Crash when
      publishing, in publisher_client_send at
      res_pjsip_outbound_publish.c (Reported by Matt Krokosz)
 * ASTERISK-25070 - Fix FTBFS on Hurd (Reported by Gabriele
      Giacone)
 * ASTERISK-26476 - chan_sip: Incorrect display option "Outbound
      reg. retry 403" in "sip show settings" (Reported by Sergey
      Grachev)
 * ASTERISK-26541 - res_pjsip_sdp_rtp: Restrict number of formats
      to maximum (Reported by Joshua Colp)
 * ASTERISK-26537 - AMI: NewConnectedLine event is not documented
      (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-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 versions (Reported
      by George Joseph)
 * ASTERISK-22480 - Embedded pjproject: build.mak contains
      hardcoded full path to version.mak (Reported by Matt Jordan)
 * ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing change
      (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 Wetzel)
 * 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-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-26480 - [patch] CLI: core set debug: Auto-completes
      File not Module (Reported by Alexander Traud)
 * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by
      Tzafrir Cohen)
 * 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-26457 - [patch] force_rport,auto_comedia: No NAT
      detection triggered. (Reported by Alexander Traud)
 * ASTERISK-26618 - build: Backport addition of librt check to
      configure.ac (Reported by Kevin Harwell)

New Features made in this release:
-----------------------------------
 * ASTERISK-26595 - ARI: Add the ability to control the source of
      video in a multi-party mixing bridge (Reported by Matt Jordan)
 * 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
      events (Reported by Matt Jordan)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.2.0

Thank you for your continued support of Asterisk!

Revision 1.4 / (download) - annotate - [select for diffs], Fri Nov 11 16:19:14 2016 UTC (2 years, 6 months ago) by jnemeth
Branch: MAIN
Changes since 1.3: +2 -2 lines
Diff to previous 1.3 (colored)

Update to Asterisk 14.1.2: this is a critical bug fix release.

The Asterisk Development Team has announced the release of Asterisk 14.1.2.

The release of Asterisk 14.1.2 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!

The following is the issue resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming
      calls after 2 minutes - rtptimeout behaving badly - regression
      (Reported by Michael Keuter)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.1.2

Thank you for your continued support of Asterisk!

Revision 1.3 / (download) - annotate - [select for diffs], Fri Oct 28 08:25:20 2016 UTC (2 years, 6 months ago) by jnemeth
Branch: MAIN
Changes since 1.2: +2 -2 lines
Diff to previous 1.2 (colored)

Update to Asterisk 14.1.1: this is a critical bug fix release.

The Asterisk Development Team has announced the release of Asterisk 14.1.1.

The release of Asterisk 14.1.1 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!

The following is the issue resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26503 - app_voicemail: Asterisk crashes when
      MailboxExists is used (Reported by Doug Lytle)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.1.1

Thank you for your continued support of Asterisk!

Revision 1.2 / (download) - annotate - [select for diffs], Thu Oct 27 06:43:39 2016 UTC (2 years, 6 months ago) by jnemeth
Branch: MAIN
Changes since 1.1: +2 -2 lines
Diff to previous 1.1 (colored)

Update to Asterisk 14.1.0: this is mostly a bug fix release.

The Asterisk Development Team has announced the release of Asterisk 14.1.0.

The release of Asterisk 14.1.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

New Features made in this release:
-----------------------------------
 * 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)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26477 - pjproject:  SEGV during SSL operations
      (Reported by George Joseph)
 * 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
      Mudgett)
 * 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 Joseph)
 * ASTERISK-26410 - core: Asterisk 14 doesn't show the header in
      the console or verbose when starting (Reported by Dan Jenkins)
 * 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-26426 - format_ogg_opus: remove from source (Reported
      by Kevin Harwell)
 * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when
      bindaddr=[::] (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-26359 - [patch] cdr_mysql: fails to use UTC if so
      instructed (Reported by Tzafrir Cohen)
 * ASTERISK-26273 - core: Won't compile when LOW_MEMORY is enabled
      (Reported by Anthony Messina)
 * ASTERISK-26391 - Consoles do not display verbose logger messages
      even when requested. (Reported by Marcelo Terres)
 * 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-19968 - TCP Session-Timers not dropping call (Reported
      by Aaron Hamstra)
 * ASTERISK-26374 - res_pjsip_multihomed: Contact port is rewritten
      for connectionful protocols (Reported by Joshua Colp)
 * 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 (Reported by
      Joshua Colp)
 * 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 Mudgett)
 * 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-26272 - chan_sip: File descriptors leak (UDP sockets)
      (Reported by Etienne Lessard)
 * ASTERISK-26264 - res_pjsip: Crash when applying ACL from
      non-existent endpoint (Reported by nappsoft)
 * 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-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-26269 - res_pjsip: Wrong state for aors without
      registered contacts after startup (Reported by nappsoft)
 * ASTERISK-26282 - AEL: macro-call in Dial application, macro
      "lacks 's' extension" (Reported by chris de rock)
 * ASTERISK-26226 - pbx: Asterisk crash on AMI action
      "ShowDialplan" when there's a circular dependency between
      contexts (Reported by Etienne Lessard)
 * ASTERISK-26299 - app_queue: Queue application sometimes stops
      calling members with Local interface (Reported by Etienne
      Lessard)
 * 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 code
      (Reported by David Brillert)
 * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with
      local channels. (Reported by Richard Mudgett)
 * ASTERISK-26288 - followme: fails to reset config items to
      default values on reload (Reported by Tzafrir Cohen)
 * 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
      abelbeck)
 * 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 Dudás)
 * ASTERISK-26305 - Asterisk 14: Two resolver unbound testsuite
      tests fail (Reported by Richard Mudgett)
 * 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
      chan_pjsip (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 Joseph)
 * 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 +
      indicate (Reported by Ross Beer)
 * ASTERISK-26183 - alembic: error when using sqlalchemy version
      1.1.0b2 (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
      res_pjsip_outbound_registration.c (Reported by Richard Mudgett)
 * ASTERISK-26265 - Errors ignored from some parts of system
      initialization. (Reported by Corey Farrell)
 * ASTERISK-26206 - [patch] res_pjsip: Use more compatible regex
      for get all (Reported by Dmitry)
 * ASTERISK-26256 - [patch] SIP/SDP origin (o=) contains brackets
      with IP6 (Reported by Alexander Traud)
 * ASTERISK-25996 - Remove "live_dangerously" requirement on
      DB(read) (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)

Improvements made in this release:
-----------------------------------
 * ASTERISK-26409 - codec_opus: Update Asterisk to support the
l
      translation codec. (Reported by Kevin Harwell)
 * ASTERISK-26289 - Announcer channels in ConfBridges cause
      inefficiencies (Reported by Mark Michelson)
 * ASTERISK-25980 - [patch]res_fax: set FAXMODE variable to let
      dialplan know what fax transport was used (Reported by Alexei
      Gradinari)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.1.0

Thank you for your continued support of Asterisk!

Revision 1.1.1.1 / (download) - annotate - [select for diffs] (vendor branch), Tue Oct 25 08:16:31 2016 UTC (2 years, 6 months ago) by jnemeth
Branch: TNF
CVS Tags: pkgsrc-base
Changes since 1.1: +0 -0 lines
Diff to previous 1.1 (colored)

     Initial import of Asterisk 14.  It has been tested to compile
and run, but not a lot of functional testing.  This does not have
the new PJSIP, which will be coming in a followup commit.  This
also does not have the patches for compiling with Clang.  For
upgrading instructions, please see:

https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+14

----- 14.0.0 -----

The Asterisk Development Team is pleased to announce the release of
Asterisk 14.0.0.

Asterisk 14 is the next major release series of Asterisk. It is a
Standard Support release, similar to Asterisk 12. For more information
about support time lines for Asterisk releases, see the Asterisk
versions page:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

For important information regarding upgrading to Asterisk 14, please
see the Asterisk wiki:
https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+14

A short list of new features includes:

* A complete overhaul of the core DNS support in Asterisk, including
  implementing full NAPTR and SRV support in the PJSIP stack via the
  libunbound library.

* The ability to publish extension state to a SIP Subscription server,
  such as Kamailio. This includes the ability to automatically generate
  a hint in the dialplan based on device state changes using the new
  autohint setting.

* Playback of media from a remote HTTP server via a URI is now supported
  by all dialplan applications and AGI. Media retrieved using a URI is
  cached in a media cache and re-used when possible.

* When using ARI to manipulate media on a resource, a list of media
  resources can now be supplied. The media resources will be played back
  sequentially in the order that they are provided.

* Channels created via ARI can now be created and handed off to Stasis
  for external control prior to performing the outbound dial. This
  enables applications to set additional state on the channel prior to
  dialing, as well as enabling certain early media scenarios.

And much more!


More information about the new features can be found on the Asterisk wiki:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Documentation

A full list of all new features can also be found in the CHANGES file:
https://github.com/asterisk/asterisk/blob/14/CHANGES

For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.0.0

Thank you for your continued support of Asterisk!

----- 14.0.1 -----

The Asterisk Development Team has announced the release of Asterisk 14.0.1.

The release of Asterisk 14.0.1 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!

The following is the issue resolved in this release:

Improvements made in this release:
-----------------------------------
 * ASTERISK-26409 - codec_opus: Update Asterisk to support the
      translation codec. (Reported by Kevin Harwell)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.0.1

Thank you for your continued support of Asterisk!

----- 14.0.2 -----

The Asterisk Development Team has announced the release of Asterisk 14.0.2.

The release of Asterisk 14.0.2 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26410 - core: Asterisk 14 doesn't show the header in
      the console or verbose when starting (Reported by Dan Jenkins)
 * ASTERISK-26426 - format_ogg_opus: remove from source (Reported
      by Kevin Harwell)
 * ASTERISK-26425 - download_externals: ignore xmlstarlet return
      code for optional element (Reported by Kevin Harwell)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.0.2

Thank you for your continued support of Asterisk!

Revision 1.1 / (download) - annotate - [select for diffs], Tue Oct 25 08:16:31 2016 UTC (2 years, 6 months ago) by jnemeth
Branch: MAIN

Initial revision

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