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CVS log for pkgsrc/comms/asterisk13/Makefile

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Revision 1.47 / (download) - annotate - [select for diffs], Wed Apr 3 00:32:28 2019 UTC (7 weeks, 3 days ago) by ryoon
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CVS Tags: HEAD
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Recursive revbump from textproc/icu

Revision 1.46 / (download) - annotate - [select for diffs], Thu Dec 13 19:51:44 2018 UTC (5 months, 1 week ago) by adam
Branch: MAIN
CVS Tags: pkgsrc-2019Q1-base, pkgsrc-2019Q1, pkgsrc-2018Q4-base, pkgsrc-2018Q4
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revbump for boost 1.69.0

Revision 1.45 / (download) - annotate - [select for diffs], Sun Dec 9 18:52:18 2018 UTC (5 months, 2 weeks ago) by adam
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revbump after updating textproc/icu

Revision 1.44 / (download) - annotate - [select for diffs], Wed Nov 14 22:21:10 2018 UTC (6 months, 1 week ago) by kleink
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Revbump after cairo 1.16.0 update.

Revision 1.43 / (download) - annotate - [select for diffs], Mon Nov 12 03:51:48 2018 UTC (6 months, 1 week ago) by ryoon
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Recursive revbump from hardbuzz-2.1.1

Revision 1.42 / (download) - annotate - [select for diffs], Mon Oct 29 17:36:57 2018 UTC (6 months, 3 weeks ago) by jperkin
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asterisk*: Fix install on SunOS.

Revision 1.41 / (download) - annotate - [select for diffs], Wed Aug 22 09:43:03 2018 UTC (9 months ago) by wiz
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CVS Tags: pkgsrc-2018Q3-base, pkgsrc-2018Q3
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Recursive bump for perl5-5.28.0

Revision 1.40 / (download) - annotate - [select for diffs], Thu Aug 16 18:54:38 2018 UTC (9 months, 1 week ago) by adam
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revbump after boost-libs update

Revision 1.39 / (download) - annotate - [select for diffs], Fri Jul 20 03:34:04 2018 UTC (10 months ago) by ryoon
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Recursive revbump from textproc/icu-62.1

Revision 1.38 / (download) - annotate - [select for diffs], Mon Jul 2 11:28:50 2018 UTC (10 months, 3 weeks ago) by darcy
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CVS Tags: pkgsrc-2018Q2-base, pkgsrc-2018Q2
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Remove redundant, commented PKGREVISION.

Revision 1.37 / (download) - annotate - [select for diffs], Sun Apr 29 21:31:29 2018 UTC (12 months, 3 weeks ago) by adam
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revbump for boost-libs update

Revision 1.36 / (download) - annotate - [select for diffs], Mon Apr 16 14:34:15 2018 UTC (13 months, 1 week ago) by wiz
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Recursive bump for new fribidi dependency in pango.

Revision 1.35 / (download) - annotate - [select for diffs], Sat Apr 14 07:34:11 2018 UTC (13 months, 1 week ago) by adam
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revbump after icu update

Revision 1.34 / (download) - annotate - [select for diffs], Mon Mar 12 11:16:12 2018 UTC (14 months, 1 week ago) by wiz
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Recursive bumps for fontconfig and libzip dependency changes.

Revision 1.33 / (download) - annotate - [select for diffs], Tue Jan 23 08:26:08 2018 UTC (16 months ago) by jnemeth
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update to Asterisk 13.19.0 -- this contains both security fixes
and general bug fixes, fixes AST-2017-005, AST-2017-006, AST-2017-007,
AST-2017-008, AST-2017-009, AST-2017-10, AST-2017-11, AST-2017-12,
AST-2017-13, and AST-2017-14 (note that a number of these only
pertain to PJSIP which isn't used in pkgsrc)

----- 13.19.0 -----

The Asterisk Development Team would like to announce the release
of Asterisk 13.19.0.

The release of Asterisk 13.19.0 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-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
      incoming INVITE Request-URI.
      (Reported by Richard Mudgett)
 * ASTERISK-27413 - Add cache_media_frames debugging option.
      (Reported by Richard Mudgett)
 * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
      endpoint identification to IP only
      (Reported by Ben Merrills)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-27531 - Compiler optimizations can break module load
      sequence.
      (Reported by abelbeck)
 * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
      Contact crashes asterisk
      (Reported by Ross Beer)
 * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
      read()
      (Reported by Abhay Gupta)
 * ASTERISK-25079 - AMI bridge of channels results in MOH not
      destroyed and robotic audio on one channel
      (Reported by Zane Conkle)
 * ASTERISK-27490 - chan_console: 'set active' fails to work
      (Reported by Tzafrir Cohen)
 * ASTERISK-24756 - ConfBridge sound_muted does not work from
      CLI or AMI
      (Reported by Thomas Frederiksen)
 * ASTERISK-25649 - Transfer application does not work with
      Local channels - documentation misleading
      (Reported by Ivan Ullmann)
 * ASTERISK-25869 - chan_sip: "rejected because extension not
      found" should be logged as a security event
      (Reported by Brian J. Murrell)
 * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
      streams
      (Reported by Wim De Vlaminck)
 * ASTERISK-24329 - Music On Hold announcement cuts intro of
      music the first time it is played
      (Reported by Thomas Frederiksen)
 * ASTERISK-19657 - Coverity Report: Fix issues for error type
      CHAR_IO
      (Reported by Matt Jordan)
 * ASTERISK-27175 - iax.conf demo peer is invalid
      (Reported by Tzafrir Cohen)
 * ASTERISK-27430 - README refers to security documents that do
      not exist.
      (Reported by Corey Farrell)
 * ASTERISK-20281 - "core set verbose" behaves strangely, can't
      alias it, cli.conf example broken
      (Reported by Tim Ringenbach at Asteria Solutions Group)
 * 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-27408 - Identify causes and fix
      pjsip/resolver/srv/failover/in_dialog/transport_tcp
      (Reported by Corey Farrell)
 * ASTERISK-18411 - Queue members with hints for state_interface
      get stuck in "In Use" state.
      (Reported by Steven T.  Wheeler)
 * ASTERISK-26131 - chan_sip: Crash Asterisk (in
      sip_request_call at chan_sip.c) by making a call to a single
      character in a dot pattern match
      (Reported by Dwayne Hubbard)
 * ASTERISK-27475 - codec_opus requires libcurl
      (Reported by Samuel For)
 * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
      not applied on reload
      (Reported by John Bigelow)
 * ASTERISK-27465 - CLI Completion Not Working
      (Reported by Ross Beer)
 * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
      Variable CDR(amaflags)=...
      (Reported by Richard Mudgett)
 * ASTERISK-27453 - RTP: Blind transfer direct media scenario
      results in one way audio.
      (Reported by Richard Mudgett)
 * ASTERISK-20643 - SIP ICE support - remove hardcoded
      limitation on SDP size, make ICE support disabled by default in
      SIP, maybe provide a better warning message
      (Reported by Roy)
 * ASTERISK-26980 - pjsip: Clean up WebRTC disables
      (Reported by abelbeck)
 * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
      flooded with unauthenticated requests
      (Reported by George Joseph)
 * ASTERISK-27454 - res_http_post: Don't require
      GMIME_MAJOR_VERSION
      (Reported by Joshua Colp)
 * ASTERISK-23735 - Transcoding makes bad choice in high-rate
      translations
      (Reported by Richard Kenner)
 * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
      message.
      (Reported by Frank Durden)
 * ASTERISK-24662 - [patch] column and row headers for Signed
      Linear format variants in output of 'core show translation' are
      ambiguous
      (Reported by Rusty Newton)
 * ASTERISK-27353 - H323 audio starts with a delay of 2
      seconds.
      (Reported by Marco Giordani)
 * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
      media
      (Reported by Kevin Harwell)
 * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
      (srflx) with Dual-Stack.
      (Reported by Alexander Traud)
 * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
      IPv6 addresses.
      (Reported by Alexander Traud)
 * ASTERISK-27435 - [patch] configure:
      pjsip_evsub_set_uas_timeout not found.
      (Reported by Alexander Traud)
 * ASTERISK-27431 - Asterisk fails to build when openssl headers
      are not installed.
      (Reported by Corey Farrell)
 * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
      (Reported by Ivan Larionov)
 * ASTERISK-27421 - RTP source learning not working with devices
      that have some clock issues
      (Reported by nappsoft)
 * ASTERISK-27361 - Attended transfer crashes in Asterisk
      13.17.2
      (Reported by Alessandro Pimenta)
 * ASTERISK-27238 - Bridging: Crash freeing a frame that's
      already been freed
      (Reported by Richard Kenner)
 * ASTERISK-27412 - core: Audiohook freeing interpolated frame
      when it shouldn't.
      (Reported by Mikhail)
 * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
      channel variable before closing the file
      (Reported by George Joseph)
 * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
      insert same ip address in "source ip address" and "destination
      ip address" fields in HEP packets
      (Reported by Max Norba)
 * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
      is equal to RemoteAddress)
      (Reported by Vasilii Rogin)
 * ASTERISK-27415 - asterisk.conf: Setting astctl without
      setting astrundir is ineffective.
      (Reported by Corey Farrell)
 * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
      (Reported by Joshua Colp)
 * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
      responses.
      (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
      machines
      (Reported by Kim youngsung)
 * ASTERISK-27391 - Regression: Deadlock between AOR named lock
      and pjproject grp lock
      (Reported by shaurya jain)
 * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
      contact read from astdb or database
      (Reported by Aaron An)
 * ASTERISK-27290 - res_pjsip: PIDF contact field has
      malformed/invalid XML
      (Reported by basildane)
 * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
      values
      (Reported by seanchann.zhou)
 * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
      enabled but not configured.
      (Reported by Alexander Traud)
 * ASTERISK-26426 - format_ogg_opus: remove from source
      (Reported by Kevin Harwell)
 * ASTERISK-27378 - Modules: Fix issues with CLI completion.
      (Reported by Corey Farrell)
 * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
      character isn't allowed any more
      (Reported by Michael Maier)
 * ASTERISK-27390 - Audit menuselect module dependencies
      (Reported by Corey Farrell)
 * ASTERISK-27389 - Optional API modules should not allow
      unload.
      (Reported by Corey Farrell)
 * ASTERISK-27369 - Bridge() dialplan application fails without
      setting BRIDGERESULT channel variable
      (Reported by James Terhune)
 * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
      documentation
      (Reported by Igor Goncharovsky)
 * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
      'imap_delete_old_greeting'
      (Reported by Anthony Messina)
 * ASTERISK-27194 - jitterbuffer: Does not handle case where
      translator returns null frame.
      (Reported by Joshua Elson)
 * ASTERISK-26639 - core: Disabling xmldoc support does not
      work. Also results in abort during Asterisk startup.
      (Reported by Mr Dini)
 * ASTERISK-27372 - ARI: Node ARI client broken in latest
      versions of 13 and 14
      (Reported by Benjamin Keith Ford)
 * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
      absence of the Expires header field with an unsubscribe action.
      (Reported by Jonathan Cloots)
 * ASTERISK-25960 - The config_hook unit test causes Asterisk to
      crash if run a second time
      (Reported by George Joseph)
 * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
      when rtp_ipv6 set to yes
      (Reported by Martin Cisárik)
 * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
      curl is loaded
      (Reported by Ronald Raikes)
 * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
      but first on SDP media level.
      (Reported by Alexander Traud)
 * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
      Assertion on un/re-load: mod.id == -1
      (Reported by Tzafrir Cohen)
 * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
      enabling with conf file option - also 'sip set debug off'
      reports debugging disabled, when it really isn't
      (Reported by Rusty Newton)
 * ASTERISK-27328 - Missing openssl dependencies in
      res_rtp_asterisk and tcptls
      (Reported by Tzafrir Cohen)
 * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
      (o=) contains local address.
      (Reported by Alexander Traud)
 * ASTERISK-27343 - Fails to build in FreeBSD due to
      sys/sysmacros.h not existing there
      (Reported by Guido Falsi)
 * ASTERISK-27340 - backtrace.c: Crash due to double-free.
      (Reported by Corey Farrell)
 * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
      stopping.
      (Reported by Alexander Traud)
 * ASTERISK-27333 - sip_to_pjsip not correctly handling
      disallow=all directive
      (Reported by Torrey Searle)

Improvements made in this release:
-----------------------------------
 * ASTERISK-24297 - cdr.c: Minor code optimizations.
      (Reported by Richard Mudgett)
 * ASTERISK-27449 - [PATCH] When failing to acquire target
      during attended transfer, display wanted extension
      (Reported by Niklas Larsson)
 * ASTERISK-27456 - app_voicemail: Add new object for
      VoicemailUserEntry
      (Reported by sungtae kim)
 * ASTERISK-27380 - ast_coredumper: allow pointing out the
      asterisk binary explicitly
      (Reported by Tzafrir Cohen)
 * ASTERISK-23556 - Compilation warning for invert.c (array
      subscript is above array bounds)
      (Reported by Marcello Ceschia)
 * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
      (Reported by Richard Mudgett)
 * ASTERISK-27335 - CDR performance needs improvement.
      (Reported by Richard Mudgett)

For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.19.0

Thank you for your continued support of Asterisk!

----- 13.18.5 -----

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-13.18.5

The security advisory is available at:

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

Thank you for your continued support of Asterisk!

----- 13.18.4 -----

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-13.18.4

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!


----- 13.18.3 -----

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-13.18.3

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

Thank you for your continued support of Asterisk!

----- 13.18.2 -----

The Asterisk Development Team would like to announce the release
of Asterisk 13.18.2.

The release of Asterisk 13.18.2 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-13.18.2

Thank you for your continued support of Asterisk!

----- 13.18.0 -----

The Asterisk Development Team would like to announce the release
of Asterisk 13.18.0.

The release of Asterisk 13.18.0 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-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
      AST_VECTOR_ADD_SORTED.
      (Reported by Corey Farrell)
 * ASTERISK-27318 - res_pjsip_mwi: uninitialized value from
      ast_strings_match
      (Reported by Corey Farrell)
 * ASTERISK-27296 - [patch] False positive busy checks when
      icalendar's recurrence-id mechanism is involved
      (Reported by Benoît Dereck-Tricot)
 * ASTERISK-27284 - Status of RFC 3323 and PJSIP
      (Reported 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
      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-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
      (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-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-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 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-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
      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-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
      asymmetric_rtp_codec=no
      (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 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-27124 - app_playback.c:say_date_generic use
      timezonename parameter
      (Reported by Holger Hans Peter Freyther)
 * 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 Jenkins)
 * 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-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-13.18.0

Thank you for your continued support of Asterisk!

----- 13.17.0 ----

The Asterisk Development Team would like to announce the release
of Asterisk 13.17.0.

The release of Asterisk 13.17.0 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-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
      sockets.
      (Reported by Louis Jocelyn Paquet)
 * ASTERISK-27060 - Comment typo format_g729.c
      (Reported by Matthew Fredrickson)
 * ASTERISK-27026 - res_ari: Crash when no ari.conf
      configuration file exists
      (Reported by Ronald Raikes)
 * 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
      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-25101 - DTLS configuration can not be specified in
      the general section - documentation
      (Reported by Ben Langfeld)
 * 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-13.17.0

Thank you for your continued support of Asterisk!

Revision 1.32 / (download) - annotate - [select for diffs], Mon Jan 1 21:18:17 2018 UTC (16 months, 3 weeks ago) by adam
Branch: MAIN
Changes since 1.31: +2 -2 lines
Diff to previous 1.31 (colored)

Revbump after boost update

Revision 1.31 / (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 1.30: +2 -2 lines
Diff to previous 1.30 (colored)

Revbump after textproc/icu update

Revision 1.30 / (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 1.29: +2 -2 lines
Diff to previous 1.29 (colored)

revbump for requiring ICU 59.x

Revision 1.29 / (download) - annotate - [select for diffs], Thu Aug 24 20:03:07 2017 UTC (21 months ago) by adam
Branch: MAIN
Changes since 1.28: +2 -1 lines
Diff to previous 1.28 (colored)

Revbump for boost update

Revision 1.28 / (download) - annotate - [select for diffs], Sun Jun 4 07:51:27 2017 UTC (23 months, 3 weeks ago) by jnemeth
Branch: MAIN
CVS Tags: pkgsrc-2017Q2-base, pkgsrc-2017Q2
Changes since 1.27: +2 -2 lines
Diff to previous 1.27 (colored)

Update to Asterisk 13.16.0:  this is mostly a bugfix release.

The Asterisk Development Team would like to announce the release
of Asterisk 13.16.0.

The release of Asterisk 13.16.0 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-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-26926 - func_speex: Crash caused by frame with no
      datalen
      (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
      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-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
      ast_str_case_hash
      (Reported by Badalian Vyacheslav)
 * 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-24712 - xmpp: starttls problem causes connection
      spew
      (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
      chdir.
      (Reported by Walter Doekes)
 * 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
      (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 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-13.16.0

Thank you for your continued support of Asterisk!

Revision 1.27 / (download) - annotate - [select for diffs], Mon May 29 20:52:37 2017 UTC (23 months, 3 weeks ago) by jnemeth
Branch: MAIN
Changes since 1.26: +2 -2 lines
Diff to previous 1.26 (colored)

Add fixes for AST-2017-002, AST-2017-003, and AST-2017-004.  Note
that the first two don't affect pkgsrc as we are using chan_sip
not PJSIP.  The last only affects users of SCCP, which is Cisco's
proprietary protocol.

----- AST-2017-002

A remote crash can be triggered by sending a SIP packet to
Asterisk with a specially crafted CSeq header and a Via
header with no branch parameter. The issue is that the
PJSIP RFC 2543 transaction key generation algorithm does
not allocate a large enough buffer. By overrunning the
buffer, the memory allocation table becomes corrupted,
leading to an eventual crash.

This issue is in PJSIP, and so the issue can be fixed
without performing an upgrade of Asterisk at all. However,
we are releasing a new version of Asterisk with the bundled
PJProject updated to include the fix.

If you are running Asterisk with chan_sip, this issue does
not affect you.

----- AST-2017-003

The multi-part body parser in PJSIP contains a logical
error that can make certain multi-part body parts attempt
to read memory from outside the allowed boundaries. A
specially-crafted packet can trigger these invalid reads
and potentially induce a crash.

The issue is within the PJSIP project and not in Asterisk.
Therefore, the problem can be fixed without upgrading
Asterisk. However, we will be releasing a new version of
Asterisk where the bundled version of PJSIP has been
updated to have the bug patched.

If you are using Asterisk with chan_sip, this issue does
not affect you.

----- AST-2017-004

A remote memory exhaustion can be triggered by sending an
SCCP packet to Asterisk system with chan_skinny enabled
that is larger than the length of the SCCP header but
smaller than the packet length specified in the header. The
loop that reads the rest of the packet doesn't detect that
the call to read() returned end-of-file before the expected
number of bytes and continues infinitely. The partial
data message logging in that tight loop causes Asterisk to
exhaust all available memory.

Revision 1.26 / (download) - annotate - [select for diffs], Sat May 13 22:39:13 2017 UTC (2 years ago) by jnemeth
Branch: MAIN
Changes since 1.25: +7 -3 lines
Diff to previous 1.25 (colored)

Update to Asterisk 13.15.0.  This is mostly a bug fix release with a few
minor enhancements.  13.14.1 was released to fix AST-2017-001.

----- 13.15.0

The Asterisk Development Team would like to announce the release of
Asterisk 13.15.0.

The release of Asterisk 13.15.0 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-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-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-26313 <https://issues.asterisk.org/jira/browse/ASTERISK-26313>]
- chan_sip : Asterisk restart seems to be required for changing encryption
option
(Reported by benasse)
- [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-26580 <https://issues.asterisk.org/jira/browse/ASTERISK-26580>]
- [patch] Error during LDAP modify action when user unregisters
(Reported by Nicholas John Koch)
- [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-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)

For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.15.0

*Thank you for your continued support of Asterisk!*

----- 13.14.0

The Asterisk Development Team has announced the release of Asterisk 13.14.0.

The release of Asterisk 13.14.0 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-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 Woolley)
 * 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-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-26672 - Crash when setting remote address on RTP
      instance (Reported by Richard Mudgett)
 * 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-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-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 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-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 - regression
      (Reported by Michael Keuter)
 * ASTERISK-26503 - app_voicemail: Asterisk crashes when
      MailboxExists is used (Reported by Doug Lytle)

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-13.14.0

Thank you for your continued support of Asterisk!

-----

Revision 1.25 / (download) - annotate - [select for diffs], Sun Apr 30 01:21:30 2017 UTC (2 years ago) by ryoon
Branch: MAIN
Changes since 1.24: +2 -2 lines
Diff to previous 1.24 (colored)

Recursive revbump from boost update

Revision 1.24 / (download) - annotate - [select for diffs], Sat Apr 22 21:03:26 2017 UTC (2 years, 1 month ago) by adam
Branch: MAIN
Changes since 1.23: +2 -2 lines
Diff to previous 1.23 (colored)

Revbump after icu update

Revision 1.23 / (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 1.22: +2 -2 lines
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Recursive revbump from fonts/harfbuzz

Revision 1.22 / (download) - annotate - [select for diffs], Mon Feb 6 13:55:09 2017 UTC (2 years, 3 months ago) by wiz
Branch: MAIN
Changes since 1.21: +2 -2 lines
Diff to previous 1.21 (colored)

Recursive bump for harfbuzz's new graphite2 dependency.

Revision 1.21 / (download) - annotate - [select for diffs], Thu Jan 19 18:52:04 2017 UTC (2 years, 4 months ago) by agc
Branch: MAIN
Changes since 1.20: +4 -4 lines
Diff to previous 1.20 (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 1.20 / (download) - annotate - [select for diffs], Sun Jan 1 16:06:06 2017 UTC (2 years, 4 months ago) by adam
Branch: MAIN
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Diff to previous 1.19 (colored)

Revbump after boost update

Revision 1.19 / (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
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Recursive revbump from textproc/icu 58.1

Revision 1.18 / (download) - annotate - [select for diffs], Sun Nov 27 08:48:18 2016 UTC (2 years, 5 months ago) by jnemeth
Branch: MAIN
Changes since 1.17: +2 -2 lines
Diff to previous 1.17 (colored)

Update to Asterisk 13.13.0:  this is mainly a bug fix release with some
minor improvements.

The Asterisk Development Team has announced the release of Asterisk 13.13.0.

The release of Asterisk 13.13.0 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-26595 - ARI: Add the ability to control the source of
      video in a multi-party mixing bridge (Reported by Matt Jordan)
 * ASTERISK-26470 - ARI: Add an 'asterisk_id' field to outgoing
      events (Reported by Matt Jordan)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26608 - Compile and link failures on OpenBSD (Reported
      by snuffy)
 * ASTERISK-26343 - ASTERISK-25951 causes issues for callerid
      manipulation through agi (Reported by Morten Tryfoss)
 * 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-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-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-26468 - ari: Bridge events stop working after this
      sequence of ARI calls (Reported by Daniele Pallastrelli)
 * ASTERISK-26311 - [patch] rtp_engine: Allow more than 32 dynamic
      payload types. (Reported by Alexander Traud)
 * ASTERISK-26549 - app_dial: When PickupChan() is used some
      channels may have incorrect device state (Reported by Joshua
      Colp)
 * ASTERISK-26541 - res_pjsip_sdp_rtp: Restrict number of formats
      to maximum (Reported by Joshua Colp)
 * 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-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-26514 - Super Awesome Company: Don't specify transport
      in pjsip.conf (Reported by Rusty Newton)
 * ASTERISK-26513 - tests/channels/pjsip/qualify/auth: Crashing
      enough to be a nuisance (Reported by Joshua Colp)
 * 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-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-26482 - [patch] chan_pjsip: segfault on already
      disconnected session (Reported by Alexei Gradinari)
 * 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-26439 - chan_rtp: Crash when originating (Reported by
      Kayode)
 * 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-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)

Improvements made in this release:
-----------------------------------
 * ASTERISK-25063 - [patch]add X.509 subject alternative name
      support to Asterisk TLS support (Reported by Maciej Szmigiero)
 * 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)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.13.0

Thank you for your continued support of Asterisk!

Revision 1.17 / (download) - annotate - [select for diffs], Fri Nov 11 15:44:16 2016 UTC (2 years, 6 months ago) by jnemeth
Branch: MAIN
Changes since 1.16: +2 -2 lines
Diff to previous 1.16 (colored)

Update the Asterisk 13.12.2: this is a critical bug fix release.

The Asterisk Development Team has announced the release of Asterisk 13.12.2.

The release of Asterisk 13.12.2 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-13.12.2

Thank you for your continued support of Asterisk!

Revision 1.16 / (download) - annotate - [select for diffs], Sat Oct 29 02:10:06 2016 UTC (2 years, 6 months ago) by jnemeth
Branch: MAIN
Changes since 1.15: +2 -2 lines
Diff to previous 1.15 (colored)

Update to Asterisk 13.12.1: this is a critical bug fix release.

The Asterisk Development Team has announced the release of Asterisk 13.12.1.

The release of Asterisk 13.12.1 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-13.12.1

Thank you for your continued support of Asterisk!

Revision 1.15 / (download) - annotate - [select for diffs], Thu Oct 27 01:08:17 2016 UTC (2 years, 6 months ago) by jnemeth
Branch: MAIN
Changes since 1.14: +2 -3 lines
Diff to previous 1.14 (colored)

Update to Asterisk 13.12.0: this is mostly a bug fix release.

The Asterisk Development Team has announced the release of Asterisk 13.12.0.

The release of Asterisk 13.12.0 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-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-26263 - SQL error when using realtime and registering
      extension / inserting into ps_contacts (Reported by Jeppe Ryskov
      Larsen)
 * 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-19968 - TCP Session-Timers not dropping call (Reported
      by Aaron Hamstra)
 * 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-26288 - followme: fails to reset config items to
      default values on reload (Reported by Tzafrir Cohen)
 * 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-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-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-26299 - app_queue: Queue application sometimes stops
      calling members with Local interface (Reported by Etienne
      Lessard)
 * 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-26269 - res_pjsip: Wrong state for aors without
      registered contacts after startup (Reported by nappsoft)
 * 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-25472 - Swagger scripts are not replacing format
      variable in file brief (Reported by Corey Farrell)
 * ASTERISK-26228 - res_pjsip_sdp_rtp: G729A does not include
      annexb=no attribute. (Reported by Ali Ghavidel)
 * ASTERISK-25984 - res_odbc relies on res_odbc_transaction, but
      it's not mandatory to compile it (Reported by József Dudás)
 * 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-26233 - pbx: Failure to remove inconsistent extension
      names (Reported by Corey Farrell)
 * ASTERISK-26164 - XMPP no longer triggers NOTIFY to device via
      chan_pjsip (Reported by Ross Beer)
 * 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-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-25797 - app_queue: Crash when calling a queue with a
      member with a forward to an nonexistent extension (Reported by
      Etienne Lessard)
 * ASTERISK-26268 - alembic: 'auth_username' not in PJSIP
      'identify_by' enum (Reported by Joshua Colp)
 * 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-26280 - DNS lookups can block channel media paths
      (Reported by Mark Michelson)
 * 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)

Improvements made in this release:
-----------------------------------
 * ASTERISK-26409 - codec_opus: Update Asterisk to support the
      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-13.12.0

Thank you for your continued support of Asterisk!

Revision 1.14 / (download) - annotate - [select for diffs], Sun Oct 9 21:41:55 2016 UTC (2 years, 7 months ago) by wiz
Branch: MAIN
Changes since 1.13: +2 -2 lines
Diff to previous 1.13 (colored)

Recursive bump for all users of pgsql now that the default is 95.

Revision 1.13 / (download) - annotate - [select for diffs], Fri Oct 7 18:25:40 2016 UTC (2 years, 7 months ago) by adam
Branch: MAIN
Changes since 1.12: +2 -1 lines
Diff to previous 1.12 (colored)

Revbump post boost update

Revision 1.12 / (download) - annotate - [select for diffs], Fri Sep 23 17:50:19 2016 UTC (2 years, 8 months ago) by jnemeth
Branch: MAIN
CVS Tags: pkgsrc-2016Q3-base, pkgsrc-2016Q3
Changes since 1.11: +9 -3 lines
Diff to previous 1.11 (colored)

Update to Asterisk 13.11.2: this is mainly a bug fix release
including two security issues:  AST-2016-006 and AST-2016-007.
Note that AST-2016-006 only affected setups using PJSIP, which
pkgsrc Asterisk does not.

pkgsrc changes:
- don't use gethostbyname_r on NetBSD
- eliminte conflict with new hmac(1) function on NetBSD

----- AST-2016-006

Asterisk can be crashed remotely by sending an ACK to it from an
endpoint username that Asterisk does not recognize.  Most SIP
request types result in an "artificial" endpoint being looked up,
but ACKs bypass this lookup. The resulting NULL pointer results in
a crash when attempting to determine if ACLs should be applied.

This issue was introduced in the Asterisk 13.10 release and only
affects that release.

This issue only affects users using the PJSIP stack with Asterisk.
Those users that use chan_sip are unaffected.

----- AST-2016-007

The overlap dialing feature in chan_sip allows chan_sip to report
to a device that the number that has been dialed is incomplete and
more digits are required. If this functionality is used with a
device that has performed username/password authentication RTP
resources are leaked.  This occurs because the code fails to release
the old RTP resources before allocating new ones in this scenario.
If all resources are used then RTP port exhaustion will occur and
no RTP sessions are able to be set up.

----- 13.11.2

The Asterisk Development Team has announced the release of Asterisk 13.11.2.

The release of Asterisk 13.11.2 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-26349 -  13.11.1 res_pjsip/pjsip_distributor.c: Request
      'REGISTER' failed (Reported by Dmitry Melekhov)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.11.2

Thank you for your continued support of Asterisk!

----- 13.11.0

The Asterisk Development Team has announced the release of Asterisk 13.11.0.

The release of Asterisk 13.11.0 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-25904 - PJSIP: add contact.updated event (Reported by
      Alexei Gradinari)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26269 - res_pjsip: Wrong state for aors without
      registered contacts after startup (Reported by nappsoft)
 * ASTERISK-26299 - app_queue: Queue application sometimes stops
      calling members with Local interface (Reported by Etienne
      Lessard)
 * ASTERISK-26148 - pjsip: Cannot compile 13.10.0-rc1:
      "libasteriskpj.so: undefined reference to..." (Reported by Hans
      van Eijsden)
 * ASTERISK-26237 - Fax is detected on regular calls. (Reported by
      Richard Mudgett)
 * ASTERISK-26227 - sqlalchemy error due to long identifier name
      (Reported by Mark Michelson)
 * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported
      by Aaron Hamstra)
 * ASTERISK-26214 - Allow arbitrary time for fax detection to end
      on a channel (Reported by Richard Mudgett)
 * ASTERISK-23013 - [patch] Deadlock between 'sip show channels'
      command and attended transfer handling (Reported by Ben
      Smithurst)
 * ASTERISK-26216 - res_fax: Deadlock when detect fax while channel
      executing Playback (Reported by Richard Mudgett)
 * ASTERISK-26212 - [patch] Makefile: Retain XML Declaration and
      DTD in docs. (Reported by Alexander Traud)
 * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in
      conditional code. (Reported by Corey Farrell)
 * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence
      number even on lost packets. (Reported by Alexander Traud)
 * ASTERISK-26038 - 'make install' doesn't seem to install OS/X
      init files (Reported by Tzafrir Cohen)
 * ASTERISK-26200 - [patch] res_pjsip_mwi: improve realtime
      performance - remove unneeded check on endpoint's contacts.
      (Reported by Alexei Gradinari)
 * ASTERISK-26133 - app_queue: Queue members receive multiple calls
      (Reported by Richard Miller)
 * ASTERISK-26196 - pbx: Time based includes can leak timezone
      string (Reported by Corey Farrell)
 * ASTERISK-26193 - chan_sip: reference leak in mwi_event_cb
      (Reported by Corey Farrell)
 * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing
      DTLS failure occurred on RTP instance (Reported by Edwin
      Vandamme)
 * ASTERISK-26191 - threadpool: Leak on duplicate taskprocessor for
      ast_threadpool_serializer_group (Reported by Corey Farrell)
 * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf.
      (Reported by Alexander Traud)
 * ASTERISK-26160 - pjsip: Updated->Reachable during qualify
      (Reported by Matt Jordan)
 * ASTERISK-25289 - Build System does not respect CFLAGS and
      CXXFLAGS when building menuselect (Reported by Jeffrey Walton)
 * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out
      of bounds and bugs (Reported by Alexei Gradinari)
 * ASTERISK-26177 - func_odbc: Database handle is kept when it
      should be released (Reported by Leandro Dardini)
 * ASTERISK-26184 - chan_sip: Reference leaks in error paths.
      (Reported by Corey Farrell)
 * ASTERISK-26181 - REF_DEBUG: Node object incorrectly logged
      during duplicate replacement (Reported by Corey Farrell)
 * ASTERISK-26180 - PJSIP: provide valid tcp nodelay option for
      reuse (Reported by Scott Griepentrog)
 * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported
      by Joshua Colp)
 * ASTERISK-26172 - res_sorcery_realtime: fix bug when successful
      sql UPDATE is treated as failed if there is no affected rows.
      (Reported by Alexei Gradinari)
 * ASTERISK-25772 - res_pjsip: Unexpected two BYE when answered
      (Reported by Dmitriy Serov)
 * ASTERISK-26099 - res_pjsip_pubsub: Crash when sending request
      due to server timeout (Reported by Ross Beer)
 * ASTERISK-26144 - Crash on loading codecs g729/g723 (Reported by
      Alexei Gradinari)
 * ASTERISK-26157 - Build:   Fix errors highlighted by GCC 6.x
      (Reported by George Joseph)
 * ASTERISK-26021 - Build codecs siren7 and siren14 for Asterisk 13
      (Reported by Daniel Denson)
 * ASTERISK-26326 - Crash when dialing MulticastRTP channel
      (Reported by George Joseph)

Improvements made in this release:
-----------------------------------
 * ASTERISK-26220 - Add support for noreturn function attributes.
      (Reported by Corey Farrell)
 * ASTERISK-22131 - Update the make dependencies script to pull,
      build, and install the correct pjproject (Reported by Matt
      Jordan)
 * ASTERISK-25471 - [patch]Add subscribe_context to res_pjsip
      (Reported by JoshE)
 * ASTERISK-26159 - res_hep: enabled by default and information
      sent to default address (Reported by Ross Beer)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.11.0

Thank you for your continued support of Asterisk!

Revision 1.11 / (download) - annotate - [select for diffs], Wed Aug 3 10:22:35 2016 UTC (2 years, 9 months ago) by adam
Branch: MAIN
Changes since 1.10: +2 -1 lines
Diff to previous 1.10 (colored)

Revbump after graphics/gd update

Revision 1.10 / (download) - annotate - [select for diffs], Sun Jul 24 06:35:50 2016 UTC (2 years, 10 months ago) by jnemeth
Branch: MAIN
Changes since 1.9: +2 -3 lines
Diff to previous 1.9 (colored)

Update to Asterisk 13.10.0:  this is mainly a bug fix release.

The Asterisk Development Team has announced the release of Asterisk 13.10.0.

The release of Asterisk 13.10.0 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-26088 - Investigate heavy memory utilization by
      res_pjsip_pubsub (Reported by Richard Mudgett)
 * ASTERISK-26011 - [patch]PJSIP: add "via_addr", "via_port",
      "call_id" to contacts (Reported by Alexei Gradinari)
 * ASTERISK-25994 - [patch]res_pjsip: module load priority
      (Reported by Alexei Gradinari)
 * ASTERISK-25931 - PJSIP: add "reg_server" to contacts. (Reported
      by Alexei Gradinari)
 * ASTERISK-25835 - Authentication using 'Username' field from
      Digest (Reported by Ross Beer)
 * ASTERISK-25930 - PJSIP: disable multi domain to improve realtime
      performace (Reported by Alexei Gradinari)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26160 - pjsip: Updated->Reachable during qualify
      (Reported by Matt Jordan)
 * ASTERISK-26177 - func_odbc: Database handle is kept when it
      should be released (Reported by Leandro Dardini)
 * ASTERISK-26099 - res_pjsip_pubsub: Crash when sending request
      due to server timeout (Reported by Ross Beer)
 * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to
      v21_details (Reported by Corey Farrell)
 * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a
      self-comparison (Reported by George Joseph)
 * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim
      generates a compile error (Reported by George Joseph)
 * ASTERISK-26128 - Alembic scripts are failing (Reported by Mark
      Michelson)
 * ASTERISK-26139 - test_res_pjsip_scheduler:  Compile failure if
      pjproject isn't installed in a system location (Reported by
      George Joseph)
 * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version.
      (Reported by Alexander Traud)
 * ASTERISK-26127 - res_pjsip_session: Crash due to race condition
      between res_pjsip_session unload and timer (Reported by Joshua
      Colp)
 * ASTERISK-26083 - ARI: Announcer channels staying around after
      playback to a bridge is finished (Reported by Per Jensen)
 * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in
      http.conf (Reported by Alexander Traud)
 * ASTERISK-26069 - Asterisk truncates To: header, dropping the
      closing '>' (Reported by Vasil Kolev)
 * ASTERISK-26097 - [patch] CLI: show maximum file descriptors
      (Reported by Alexander Traud)
 * ASTERISK-25262 - Memory leak when a caller channel does multiple
      dials and CEL is enabled (Reported by Etienne Lessard)
 * ASTERISK-26092 - [Segfault] in res_rtp_asterisk.c:4268 after
      Remotely bridged channels (Reported by Niklas Larsson)
 * ASTERISK-26096 - res_hep: Crash when configuration file is
      missing (Reported by Niklas Larsson)
 * ASTERISK-26089 - Invalid security events during boot using PJSIP
      Realtime (Reported by Scott Griepentrog)
 * ASTERISK-26074 - res_odbc: Deadlock within UnixODBC (Reported by
      Ross Beer)
 * ASTERISK-26054 - Asterisk crashes (core dump) (Reported by B.
      Davis)
 * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling
      against libsrtp-1.5.0 (Reported by Patrick Laimbock)
 * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar
      cr (Reported by Alexander Traud)
 * ASTERISK-26070 - ari/channels:  Creating a local channel without
      an originator adds all audio formats to it's capabilities
      (Reported by George Joseph)
 * ASTERISK-26078 - core: Memory leak in logging (Reported by
      Etienne Lessard)
 * ASTERISK-26065 - chan_pjsip: MWI NOTIFY contents not ordered
      properly (Reported by Ross Beer)
 * ASTERISK-26063 - ${PJSIP_HEADER(read,Call-ID)} does not work -
      documentation needs clarification for when read/write is
      possible (Reported by Private Name)
 * ASTERISK-25777 - data race in threadpool (Reported by Badalian
      Vyacheslav)
 * ASTERISK-26038 - 'make install' doesn't seem to install OS/X
      init files (Reported by Tzafrir Cohen)
 * ASTERISK-26029 - parking: ast_parking_park_call should return
      parking_space instead of parking_exten (Reported by Diederik de
      Groot)
 * ASTERISK-25938 - res_odbc: MySQL/MariaDB statement
      LAST_INSERT_ID() always returns zero. (Reported by Edwin
      Vandamme)
 * ASTERISK-25941 - chan_pjsip: Crash on an immediate SIP final
      response (Reported by Javier Riveros )
 * ASTERISK-26014 - res_sorcery_astdb: Make tolerant of unknown
      fields (Reported by Joshua Colp)
 * ASTERISK-24986 - keepalive INFO packages ignored by asterisk
      (Reported by Ilya Trikoz)
 * ASTERISK-26034 - T.38 passthrough problem behind firewall due to
      early nosignal packet (Reported by George Joseph)
 * ASTERISK-26030 - call cut because of double Session-Expires
      header in re-invite after proxy authentication is required
      (Reported by George Joseph)
 * ASTERISK-25964 - Outbound registrations created via ARI/push
      configuration do not clean up outbound registrations currently
      in flight (Reported by Matt Jordan)
 * ASTERISK-26005 - res_pjsip: Multiple SIP messages are combined
      into 1 TCP packet (Reported by Ross Beer)
 * ASTERISK-25352 - res_hep_rtcp correlation_id is different then
      res_hep (Reported by Kevin Scott Adams)
 * ASTERISK-26008 - app_followme does not delete recorded name
      prompt (Reported by Tzafrir Cohen)
 * ASTERISK-26007 - res_pjsip: Endpoints deleting early after
      upgrade from 13.8.2 to 13.9 (Reported by Greg Siemon)
 * ASTERISK-25990 - PJSIP TLS registration should respect
      client_uri scheme when generating Contact URI (Reported by
      Sebastian Damm)
 * ASTERISK-25978 - res_pjsip_authenticator_digest: Should not use
      source port in nonce verification (Reported by Mark Michelson)
 * ASTERISK-25993 - pjproject: Allow bundling to not require
      everything it does (Reported by Joshua Colp)
 * ASTERISK-25956 - Compilation error in conditionally compiled
      code in config_options.c (Reported by Chris Trobridge)
 * ASTERISK-25998 - file: Crash when using nativeformats (Reported
      by Joshua Colp)
 * ASTERISK-25826 - PJSIP / Sorcery slow load from realtime
      (Reported by Ross Beer)
 * ASTERISK-25968 - pjproject_bundled:  Configure and make need to
      be re-tested (Reported by George Joseph)
 * ASTERISK-24463 - Voicemail email address corrupt or not sent
      when message is in the process of being recorded during reload
      (Reported by John Campbell)
 * ASTERISK-25970 - Segfault in pjsip_url_compare (Reported by
      Dmitriy Serov)
 * ASTERISK-25963 - func_odbc requires reconnect checks for stale
      connections (Reported by Ross Beer)
 * ASTERISK-25961 - tests/channels/SIP/sip_tls_call: Sporadic crash
      when running test (Reported by Joshua Colp)
 * ASTERISK-16115 - [patch] problem with ringinuse=no, queue
      members receive sometimes two calls (Reported by nik600)
 * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir
      only works if you manually add secret.conf yourself (Reported by
      Jonathan R. Rose)
 * ASTERISK-25950 - [patch]SIP channel does not send PeerStatus
      events for autocreated peers (Reported by Kirill Katsnelson)
 * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions
      are case sensitive to QueueName (Reported by Javier Acosta)

New Features made in this release:
-----------------------------------
 * ASTERISK-25904 - PJSIP: add contact.updated event (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-13.10.0

Thank you for your continued support of Asterisk!

Revision 1.9 / (download) - annotate - [select for diffs], Sat Jul 9 06:37:55 2016 UTC (2 years, 10 months ago) by wiz
Branch: MAIN
Changes since 1.8: +2 -1 lines
Diff to previous 1.8 (colored)

Bump PKGREVISION for perl-5.24.0 for everything mentioning perl.

Revision 1.8 / (download) - annotate - [select for diffs], Thu Jun 9 04:41:48 2016 UTC (2 years, 11 months ago) by jnemeth
Branch: MAIN
CVS Tags: pkgsrc-2016Q2-base, pkgsrc-2016Q2
Changes since 1.7: +2 -2 lines
Diff to previous 1.7 (colored)

Upgrade to Asterisk 13.9.1: this is a bugfix release.  Note that
since the package doesn't support PJSIP (yet), all reference to
PJSIP bugs are not applicable.

----- 13.9.1

The Asterisk Development Team has announced the release of Asterisk 13.9.1.

The release of Asterisk 13.9.1 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-26007 - res_pjsip: Endpoints deleting early after
      upgrade from 13.8.2 to 13.9 (Reported by Greg Siemon)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.9.1

Thank you for your continued support of Asterisk!

----- 13.9.0

The Asterisk Development Team has announced the release of Asterisk 13.9.0.

The release of Asterisk 13.9.0 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:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-25963 - func_odbc requires reconnect checks for stale
      connections (Reported by Ross Beer)
 * ASTERISK-25970 - Segfault in pjsip_url_compare (Reported by
      Dmitriy Serov)
 * ASTERISK-25938 - res_odbc: MySQL/MariaDB statement
      LAST_INSERT_ID() always returns zero. (Reported by Edwin
      Vandamme)
 * ASTERISK-25927 - Removed option "registertrying" is still
      documented in sip.conf.sample (Reported by Etienne Lessard)
 * ASTERISK-25947 - Protocol transfers to stasis applications are
      missing the StasisStart with the replace_channel object.
      (Reported by Richard Mudgett)
 * ASTERISK-24649 - Pushing of channel into bridge fails; Stasis
      fails to get app name (Reported by John Bigelow)
 * ASTERISK-24782 - StasisEnd event not present for channel that
      was swapped out for another after completing attended transfer
      (Reported by John Bigelow)
 * ASTERISK-25942 - res_pjsip_caller_id: Transfer results in mixed
      ConnectedLine information (Reported by George Joseph)
 * ASTERISK-25928 - res_pjsip: URI validation done outside of PJSIP
      thread (Reported by Joshua Colp)
 * ASTERISK-25929 - res_pjsip_registrar: AOR_CONTACT_ADDED events
      not raised (Reported by Joshua Colp)
 * ASTERISK-25934 - chan_sip should not require sipregs or
      updateable sippeers table unless rt (Reported by Jaco Kroon)
 * ASTERISK-25888 - Frequent segfaults in function can_ring_entry()
      of app_queue.c (Reported by Sébastien Couture)
 * ASTERISK-25796 - res_pjsip: DOS/Crash when TCP/TLS sockets
      exceed pjproject PJ_IOQUEUE_MAX_HANDLES (Reported by George
      Joseph)
 * ASTERISK-25707 - Long contact URIs or hostnames can crash
      pjproject/Asterisk under certain conditions (Reported by George
      Joseph)
 * ASTERISK-25123 - Bracketed IPv6 Contact header parameter
      unparsable with Asterisk/PJSIP (Reported by Anthony Messina)
 * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
 * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE
      without adding them to the local hangupcauses via
      ast_channel_hangupcause_hash_set (Reported by Jaco Kroon)
 * ASTERISK-25885 - res_pjsip: Race condition between adding
      contact and automatic expiration (Reported by Joshua Colp)
 * ASTERISK-25910 - pjproject:  Via headers are not parsed when
      "received" contains an IPv6 address (Reported by George Joseph)
 * ASTERISK-25890 - Asterisk 13.8.0 alembic database update fails
      (Reported by Harley Peters)
 * ASTERISK-25894 - [patch] webrtc video broken due to missing
      marker bits in RTP streams (Reported by Jacek Konieczny)
 * ASTERISK-25854 - No audio after HOLD/RESUME - incorrect
      a=recvonly in SDP from Asterisk (Reported by Robert McGilvray)
 * ASTERISK-25873 - res_pjsip: Bundled pjproject: compile error,
      cannot find -lasteriskpj (Reported by Hans van Eijsden)
 * ASTERISK-25882 - ARI: Crash can occur due to race condition when
      attempting to operate on a hung up channel (Part 2) (Reported by
      Richard Mudgett)
 * ASTERISK-25867 - [patch] Video delay on app_echo (Reported by
      Jacek Konieczny)
 * ASTERISK-24605 - res_parking option parkeddynamic does not work
      with the core Features 'parkcall' (DTMF initiated parking)
      (Reported by Philip Correia)
 * ASTERISK-25826 - PJSIP / Sorcery slow load from realtime
      (Reported by Ross Beer)
 * ASTERISK-24596 - Unclear how to use Park application with
      res_parking 'parkeddynamic' enabled. Documentation? (Reported by
      Philip Correia)
 * ASTERISK-24543 - Asterisk 13 responds to SIP Invite with all
      possible codecs configured for peer as opposed to intersection
      of configured codecs and offered codecs (Reported by Taylor
      Hawkes)
 * ASTERISK-25825 - Crashes during shutdown when running CLI
      commands (Reported by Mark Michelson)
 * ASTERISK-25407 - Asterisk fails to log to multiple syslog
      destinations (Reported by Elazar Broad)
 * ASTERISK-25510 - [patch]Log to syslog failing (Reported by
      Michael Newton)
 * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
      data corruption (Reported by Gianluca Merlo)

Improvements made in this release:
-----------------------------------
 * ASTERISK-25865 - Message-Account Missing From PJSIP MWI
      (Reported by Ross Beer)
 * ASTERISK-25444 - [patch]Music On Hold Warning misleading
      (Reported by Conrad de Wet)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.9.0

Thank you for your continued support of Asterisk!

Revision 1.7 / (download) - annotate - [select for diffs], Fri May 6 07:41:06 2016 UTC (3 years ago) by jnemeth
Branch: MAIN
Changes since 1.6: +9 -9 lines
Diff to previous 1.6 (colored)

Update to Asterisk 13.8.2: this is mainly a bug fix release.  It
also contains fixes for AST-2016-004 and AST-2016-005.  However,
those issues only affected the pjsip module.  Since Asterisk in
pkgsrc doesn't (yet) use pjsip, it wasn't affected.

----- 13.8.2

The Asterisk Development Team has announced the release of Asterisk 13.8.2.

The release of Asterisk 13.8.2 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-25929 - res_pjsip_registrar: AOR_CONTACT_ADDED events
      not raised (Reported by Joshua Colp)
 * ASTERISK-25928 - res_pjsip: URI validation done outside of PJSIP
      thread (Reported by Joshua Colp)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.8.2

Thank you for your continued support of Asterisk!

----- 13.8.0

The Asterisk Development Team has announced the release of Asterisk 13.8.0.

The release of Asterisk 13.8.0 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-24919 - res_pjsip_config_wizard: Ability to write
      contents to file (Reported by Ray Crumrine)
 * ASTERISK-25670 - Add regcontext to PJSIP (Reported by Daniel
      Journo)
 * ASTERISK-25480 - [patch]Add field PauseReason on
      QueueMemberStatus (Reported by Rodrigo Ramirez Norambuena)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-25849 - chan_pjsip: transfers with direct media
      sometimes drops audio (Reported by Kevin Harwell)
 * ASTERISK-25113 - install_prereq in Debian 8 without "standard
      system utilities" (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-25814 - Segfault at f ip in res_pjsip_refer.so
      (Reported by Sergio Medina Toledo)
 * ASTERISK-25023 - Deadlock in chan_sip in
      update_provisional_keepalive (Reported by Arnd Schmitter)
 * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
      channel (Reported by Filip Frank)
 * ASTERISK-25829 - res_pjsip: PJSIP does not accept spaces when
      separating multiple AORs (Reported by Mateusz Kowalski)
 * ASTERISK-25771 - ARI:Crash - Attended transfers of channels into
      Stasis application. (Reported by Javier Riveros )
 * ASTERISK-25830 - Revision 2451d4e breaks NAT (Reported by Sean
      Bright)
 * ASTERISK-25582 - Testsuite: Reactor timeout error in
      tests/fax/pjsip/directmedia_reinvite_t38 (Reported by Matt
      Jordan)
 * ASTERISK-25811 - Unable to delete object from sorcery cache
      (Reported by Ross Beer)
 * ASTERISK-25800 - [patch] Calculate talktime when is first call
      answered (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-25727 - RPM build requires OPTIONAL_API cflag due to
      PJSIP requirement (Reported by Gergely Dömsödi)
 * ASTERISK-25337 - Crash on PJSIP_HEADER Add P-Asserted-Identity
      when calling from Gosub (Reported by Jacques Peacock)
 * ASTERISK-25738 - res_pjsip_pubsub: Crash while executing
      OutboundSubscriptionDetail ami action (Reported by Kevin
      Harwell)
 * ASTERISK-25721 - [patch] res_phoneprov: memory leak and
      heap-use-after-free (Reported by Badalian Vyacheslav)
 * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
      returns garbage (Reported by Etienne Lessard)
 * ASTERISK-25751 - res_pjsip: Support
      pjsip_dlg_create_uas_and_inc_lock (Reported by Joshua Colp)
 * ASTERISK-25606 - Core dump when using transports in sorcery
      (Reported by Martin Mouka)
 * ASTERISK-20987 - non-admin users, who join muted conference are
      not being muted (Reported by hristo)
 * ASTERISK-25737 - res_pjsip_outbound_registration: line option
      not in Alembic (Reported by Joshua Colp)
 * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
      udptl_rx_packet cause ast_frdup crash (Reported by Walter
      Doekes)
 * ASTERISK-25742 - Secondary IFP Packets can result in accessing
      uninitialized pointers and a crash (Reported by Torrey Searle)
 * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
      Vulnerability - Investigate vulnerability of HTTP server
      (Reported by Alex A. Welzl)
 * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
      non-default timert1 (Reported by Alexander Traud)
 * ASTERISK-25702 - PjSip realtime DB and Cache Errors since
      upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported by
      Nic Colledge)
 * ASTERISK-25730 - build:  make uninstall after make distclean
      tries to remove root (Reported by George Joseph)
 * ASTERISK-25725 - core: Incorrect XML documentation may result in
      weird behavior (Reported by Joshua Colp)
 * ASTERISK-25722 - ASAN & testsute: stack-buffer-overflow in
      sip_sipredirect (Reported by Badalian Vyacheslav)
 * ASTERISK-25709 - ARI: Crash can occur due to race condition when
      attempting to operate on a hung up channel (Reported by Mark
      Michelson)
 * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
      by Badalian Vyacheslav)
 * ASTERISK-25685 - infrastructure: Run alembic in Jenkins build
      script (Reported by Joshua Colp)
 * ASTERISK-25712 - Second call to already-on-call phone and
      Asterisk sends "Ready" (Reported by Richard Mudgett)
 * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
      (Reported by Badalian Vyacheslav)
 * ASTERISK-25179 - CDR(billsec,f) and CDR(duration,f) report
      incorrect values (Reported by Gianluca Merlo)
 * ASTERISK-25611 - core: threadpool thread_timeout_thrash unit
      test sporadically failing (Reported by Joshua Colp)
 * ASTERISK-24097 - Documentation - CHANNEL function help text
      missing 'linkedid' argument (Reported by Steven T. Wheeler)
 * ASTERISK-25700 - main/config: Clean config maps on shutdown.
      (Reported by Corey Farrell)
 * ASTERISK-25696 - bridge_basic: don't cache xferfailsound during
      a transfer (Reported by Kevin Harwell)
 * ASTERISK-25697 - bridge_basic: don't play an attended transfer
      fail sound after target hangs up (Reported by Kevin Harwell)
 * ASTERISK-25683 - res_ari: Asterisk fails to start if compiled
      with MALLOC_DEBUG  (Reported by yaron nahum)
 * ASTERISK-25686 - PJSIP: qualify_timeout is a double, database
      schema is an integer (Reported by Marcelo Terres)
 * ASTERISK-25690 - Hanging up when executing connected line sub
      does not cause hangup (Reported by Joshua Colp)
 * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
      reload' cause a crash (Reported by Sean Bright)
 * ASTERISK-25632 - res_pjsip_sdp_rtp: RTP is sent from wrong IP
      address when multihomed (Reported by Olivier Krief)
 * ASTERISK-25637 - Multi homed server using wrong IP (Reported by
      Daniel Journo)
 * ASTERISK-25394 - pbx: Incorrect device and presence state when
      changing hint details (Reported by Joshua Colp)
 * ASTERISK-25640 - pbx: Deadlock on features reload and state
      change hint. (Reported by Krzysztof Trempala)
 * ASTERISK-25681 - devicestate: Engine thread is not shut down
      (Reported by Corey Farrell)
 * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
      shutdown (Reported by Corey Farrell)
 * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
      Corey Farrell)
 * ASTERISK-25675 - Endpoint not listed as Unreachable (Reported by
      Daniel Journo)
 * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
      by Corey Farrell)
 * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
      Farrell)
 * ASTERISK-25668 - res_pjsip: Deadlock in distributor (Reported by
      Mark Michelson)
 * ASTERISK-25664 - ast_format_cap_append_by_type leaks a reference
      (Reported by Corey Farrell)
 * ASTERISK-25647 - bug of cel_radius.c: wrong point of
      ADD_VENDOR_CODE (Reported by Aaron An)
 * ASTERISK-25317 - asterisk sends too many stun requests (Reported
      by Stefan Engström)
 * ASTERISK-25137 - endpoint stasis messages are delivered twice
      (Reported by Vitezslav Novy)
 * ASTERISK-25116 - res_pjsip:  Two PeerStatus AMI messages are
      sent for every status change (Reported by George Joseph)
 * ASTERISK-25641 - bridge: GOTO_ON_BLINDXFR doesn't work on
      transfer initiated channel (Reported by Dmitry Melekhov)
 * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
      Brandon)
 * ASTERISK-25442 - using realtime (mysql) queue members are never
      updated in wait_our_turn function (app_queue.c)  (Reported by
      Carlos Oliva)
 * ASTERISK-25625 - res_sorcery_memory_cache: Add full backend
      caching (Reported by Joshua Colp)
 * ASTERISK-25601 - json: Audit reference usage and thread safety
      (Reported by Joshua Colp)
 * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
      sungtae kim)

Improvements made in this release:
-----------------------------------
 * ASTERISK-25495 - [patch] Prevent old-update packages on
      repository Debian systems (Reported by Rodrigo Ramirez
      Norambuena)
 * ASTERISK-25846 - Gracefully deal with Absent Stasis Apps
      (Reported by Andrew Nagy)
 * ASTERISK-25791 - res_pjsip_caller_id: Lack of support for
      Anonymous <anonymous at anonymous.invalid> (Reported by Anthony
      Messina)
 * ASTERISK-24813 - asterisk.c: #if statement in listener()
      confuses code folding editors (Reported by Corey Farrell)
 * ASTERISK-25767 - [patch] Add check to configure for sanitizes
      (Reported by Badalian Vyacheslav)
 * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
      core set (Reported by Rusty Newton)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.8.0

Thank you for your continued support of Asterisk!

Revision 1.6 / (download) - annotate - [select for diffs], Mon Apr 11 19:01:44 2016 UTC (3 years, 1 month ago) by ryoon
Branch: MAIN
Changes since 1.5: +2 -2 lines
Diff to previous 1.5 (colored)

Recursive revbump from textproc/icu 57.1

Revision 1.5 / (download) - annotate - [select for diffs], Sat Mar 5 11:28:10 2016 UTC (3 years, 2 months ago) by jperkin
Branch: MAIN
CVS Tags: pkgsrc-2016Q1-base, pkgsrc-2016Q1
Changes since 1.4: +2 -1 lines
Diff to previous 1.4 (colored)

Bump PKGREVISION for security/openssl ABI bump.

Revision 1.4 / (download) - annotate - [select for diffs], Thu Feb 25 11:32:19 2016 UTC (3 years, 2 months ago) by jperkin
Branch: MAIN
Changes since 1.3: +3 -5 lines
Diff to previous 1.3 (colored)

Use OPSYSVARS.

Revision 1.3 / (download) - annotate - [select for diffs], Sun Feb 7 09:13:34 2016 UTC (3 years, 3 months ago) by jnemeth
Branch: MAIN
Changes since 1.2: +20 -18 lines
Diff to previous 1.2 (colored)

Update Asterisk to 13.7.2: this is mainly bug fixes with some minor
features and fixes for AST-2016-001, AST-2016-002, and AST-2016-003.
Also some pkglinting.

----- 13.7.2

The Asterisk Development Team has announced the release of Asterisk 13.7.2.

The release of Asterisk 13.7.2 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-25702 - PjSip realtime DB and Cache Errors since
      upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported by
      Nic Colledge)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.7.2

Thank you for your continued support of Asterisk!

----- 13.7.1

The Asterisk Development Team has announced security releases for Certified
Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.

The release of these versions resolves the following security vulnerabilities:

* AST-2016-001: BEAST vulnerability in HTTP server

  The Asterisk HTTP server currently has a default configuration which allows
  the BEAST vulnerability to be exploited if the TLS functionality is enabled.
  This can allow a man-in-the-middle attack to decrypt data passing through it.

* AST-2016-002: File descriptor exhaustion in chan_sip

  Setting the sip.conf timert1 value to a value higher than 1245 can cause an
  integer overflow and result in large retransmit timeout times. These large
  timeout values hold system file descriptors hostage and can cause the system
  to run out of file descriptors.

* AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.

  If no UDPTL packets are lost there is no problem. However, a lost packet
  causes Asterisk to use the available error correcting redundancy packets. If
  those redundancy packets have zero length then Asterisk uses an uninitialized
  buffer pointer and length value which can cause invalid memory accesses later
  when the packet is copied.

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

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1

The security advisories are available at:

 * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
 * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
 * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf

Thank you for your continued support of Asterisk!

----- 13.7.0

The Asterisk Development Team has announced the release of Asterisk 13.7.0.

The release of Asterisk 13.7.0 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-25419 - Dialplan Application for Integration of StatsD
      (Reported by Ashley Sanders)
 * ASTERISK-25549 - Confbridge: Add participant timeout option
      (Reported by Mark Michelson)
 * ASTERISK-24922 - ARI: Add the ability to intercept hold and
      raise an event (Reported by Matt Jordan)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-25689 - pjsip show contacts not working in Asterisk
      13.7rc2 (Reported by Marcelo Terres)
 * ASTERISK-25640 - pbx: Deadlock on features reload and state
      change hint. (Reported by Krzysztof Trempala)
 * ASTERISK-25664 - ast_format_cap_append_by_type leaks a reference
      (Reported by Corey Farrell)
 * ASTERISK-25601 - json: Audit reference usage and thread safety
      (Reported by Joshua Colp)
 * ASTERISK-25625 - res_sorcery_memory_cache: Add full backend
      caching (Reported by Joshua Colp)
 * ASTERISK-25615 - res_pjsip: Setting transport async_operations >
      1 causes segfault on tls transports (Reported by George Joseph)
 * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
      thread of asterisk is not released (Reported by Hiroaki Komatsu)
 * ASTERISK-25619 - res_chan_stats not sending the correct
      information to StatsD (Reported by Tyler Cambron)
 * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
      Corey Farrell)
 * ASTERISK-25609 - [patch]Asterisk may crash when calling
      ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
 * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
      answer waiting time is more than ~7sec (Reported by Aleksei
      Kulakov)
 * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
      (Reported by Alexander Traud)
 * ASTERISK-25616 - Warning with a Codec Module which supports PLC
      with FEC (Reported by Alexander Traud)
 * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
      Dudás József)
 * ASTERISK-25608 - res_pjsip/contacts/statsd:  Lifecycle events
      aren't consistent (Reported by George Joseph)
 * ASTERISK-25584 - [patch] format-attribute module: VP8 missing
      (Reported by Alexander Traud)
 * ASTERISK-25583 - [patch] format-attribute module: RFC 7587 (Opus
      Codec) (Reported by Alexander Traud)
 * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
      that module installed (Reported by Ben Langfeld)
 * ASTERISK-25595 - Unescaped : in messge sent to statsd (Reported
      by Niklas Larsson)
 * ASTERISK-25476 - chan_sip loses registrations after a while
      (Reported by Michael Keuter)
 * ASTERISK-25598 - res_pjsip:  Contact status messages are
      printing a hash instead of the uri (Reported by George Joseph)
 * ASTERISK-25600 - bridging: Inconsistency in BRIDGEPEER (Reported
      by Jonathan Rose)
 * ASTERISK-25582 - Testsuite: Reactor timeout error in
      tests/fax/pjsip/directmedia_reinvite_t38 (Reported by Matt
      Jordan)
 * ASTERISK-25593 - fastagi: record file closed after sending
      result (Reported by Kevin Harwell)
 * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
      it's assumed to (Reported by Walter Doekes)
 * ASTERISK-25590 - CLI Usage info for 'pjsip send notify'
      references incorrect config (Reported by Corey Farrell)
 * ASTERISK-25165 - Testsuite - Sorcery memory cache leaks
      (Reported by Corey Farrell)
 * ASTERISK-25575 - res_pjsip: Dynamic outbound registrations
      created via ARI are not loaded into memory on Asterisk
      start/restart (Reported by Matt Jordan)
 * ASTERISK-25545 - [patch] translation module gets cached not
      joint format (Reported by Alexander Traud)
 * ASTERISK-25573 - [patch] H.264 format attribute module: resets
      whole SDP (Reported by Alexander Traud)
 * ASTERISK-24958 - Forwarding loop detection inhibits certain
      desirable scenarios (Reported by Mark Michelson)
 * ASTERISK-25561 - app_queue.c line 6503 (try_calling): mutex
      'qe->chan' freed more times than we've locked! (Reported by Alec
      Davis)
 * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
      Joshua Colp)
 * ASTERISK-25160 - [patch] Opus Codec: SIP/SDP line fmtp missing
      when called internally (Reported by Alexander Traud)
 * ASTERISK-25535 - [patch] format creation on module load instead
      of cache (Reported by Alexander Traud)
 * ASTERISK-25449 - main/sched: Regression introduced by
      5c713fdf18f causes erroneous duplicate RTCP messages; other
      potential scheduling issues in chan_sip/chan_skinny (Reported by
      Matt Jordan)
 * ASTERISK-25546 - threadpool: Race condition between idle timeout
      and activation (Reported by Joshua Colp)
 * ASTERISK-25537 - [patch] format-attribute module: RFC or
      internal defaults? (Reported by Alexander Traud)
 * ASTERISK-25533 - [patch] buffer for ast_format_cap_get_names
      only 64 bytes (Reported by Alexander Traud)
 * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
      the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
      Doekes)
 * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
      Walter Doekes)
 * ASTERISK-24779 - Passthrough OPUS codec not working with
      chan_pjsip (Reported by PowerPBX)
 * ASTERISK-25522 - ARI: Crash when creating channel via ARI
      originate with requesting channel (Reported by Matt Jordan)
 * ASTERISK-25434 - Compiler flags not reported in 'core show
      settings' despite usage during compilation (Reported by Rusty
      Newton)
 * ASTERISK-24106 - WebSockets Automatically decides what driver it
      will use  (Reported by Andrew Nagy)
 * ASTERISK-25513 - Crash: malloc failed with high load of
      subscriptions. (Reported by John Bigelow)
 * ASTERISK-25505 - res_pjsip_pubsub: Crash on off-nominal when UAS
      dialog can't be created (Reported by Joshua Colp)
 * ASTERISK-24543 - Asterisk 13 responds to SIP Invite with all
      possible codecs configured for peer as opposed to intersection
      of configured codecs and offered codecs (Reported by Taylor
      Hawkes)
 * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
      bounds and missing paren issues (Reported by George Joseph)
 * ASTERISK-25485 - res_pjsip_outbound_registration: registration
      stops due to 400 response (Reported by Kevin Harwell)
 * ASTERISK-25486 - res_pjsip: Fix deadlock when validating URIs
      (Reported by Joshua Colp)
 * ASTERISK-7803 - [patch] Update the maximum packetization values
      in frame.c (Reported by dea)
 * ASTERISK-25484 - [patch] autoframing=yes has no effect (Reported
      by Alexander Traud)
 * ASTERISK-25461 - Nested dialplan #includes don't work as
      expected. (Reported by Richard Mudgett)
 * ASTERISK-25455 - Deadlock of PJSIP realtime over
      res_config_pgsql  (Reported by mdu113)
 * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
      (Reported by Olle Johansson)
 * ASTERISK-25435 - Asterisk periodically hangs. UDP Recv-Q greatly
      exceeds zero. (Reported by Dmitriy Serov)
 * ASTERISK-25451 - Broken video - erased rtp marker bit (Reported
      by Stefan Engström)
 * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
      exist in AstDB (Reported by Andrew Nagy)
 * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
      parsing (Reported by ffs)
 * ASTERISK-25404 - segfault/crash in chan_pjsip_hangup ... at
      chan_pjsip.c (Reported by Chet Stevens)
 * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
      (Reported by Bojan Nemi)
 * ASTERISK-25441 - Deadlock in res_sorcery_memory_cache. (Reported
      by Richard Mudgett)
 * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
      ICE is not enabled (Reported by Joshua Colp)

Improvements made in this release:
-----------------------------------
 * ASTERISK-25618 - res_pjsip:  Check for readability of TLS files
      at startup (Reported by George Joseph)
 * ASTERISK-25572 - Endpoints: Add StatsD stats for Asterisk
      endpoints (Reported by Matt Jordan)
 * ASTERISK-25571 - PJSIP: Add StatsD stats for some common PJSIP
      objects (Reported by Matt Jordan)
 * ASTERISK-25518 - taskprocessor: Add high water mark (Reported by
      Jonathan Rose)
 * ASTERISK-25477 - pjsip show "command" like [criteria] (Reported
      by Bryant Zimmerman)
 * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
      configure (Reported by Badalian Vyacheslav)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.7.0

Thank you for your continued support of Asterisk!

Revision 1.2 / (download) - annotate - [select for diffs], Sat Dec 5 23:42:44 2015 UTC (3 years, 5 months ago) by jnemeth
Branch: MAIN
CVS Tags: pkgsrc-2015Q4-base, pkgsrc-2015Q4
Changes since 1.1: +1 -1 lines
Diff to previous 1.1 (colored)

     Initial import of Asterisk 13.  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+13

-----

The Asterisk Development Team is pleased to announce the release
of Asterisk 13.0.0.

Asterisk 13 is the next major release series of Asterisk. It is a
Long Term Support (LTS) release, similar to Asterisk 11. 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 13, please
see the Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+13

A short list of new features includes:

* Asterisk security events are now provided via AMI, allowing end
  users to monitor their Asterisk system in real time for security
  related issues.

* Both AMI and ARI now allow external systems to control the state
  of a mailbox.  Using AMI actions or ARI resources, external
  systems can programmatically trigger Message Waiting Indicators
  (MWI) on subscribed phones. This is of particular use to those
  who want to build their own VoiceMail application using ARI.

* ARI now supports the reception/transmission of out of call text
  messages using any supported channel driver/protocol stack through
  ARI. Users receive out of call text messages as JSON events over
  the ARI websocket connection, and can send out of call text
  messages using HTTP requests.

* The PJSIP stack now supports RFC 4662 Resource Lists, allowing
  Asterisk to act as a Resource List Server. This includes defining
  lists of presence state, mailbox state, or lists of presence
  state/mailbox state; managing subscriptions to lists; and batched
  delivery of NOTIFY requests to subscribers.

* The PJSIP stack can now be used as a means of distributing device
  state or mailbox state via PUBLISH requests to other Asterisk
  instances.  This is analogous to Asterisk's clustering support
  using XMPP or Corosync; unlike existing clustering mechanisms,
  using the PJSIP stack to perform the distribution of state does
  not rely on another daemon or server to perform the work.

And much more!

More information about the new features can be found on the Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Documentation

A full list of all new features can also be found in the CHANGES file:

http://svnview.digium.com/svn/asterisk/branches/13/CHANGES

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

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.0

Thank you for your continued support of Asterisk!

-----

The Asterisk Development Team has announced the release of Asterisk 13.1.0.

The release of Asterisk 13.1.0 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-24554 - AMI/ARI: Generate events on connected line
      changes (Reported by Matt Jordan)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling
      against libsrtp-1.5.0 (Reported by Patrick Laimbock)
 * ASTERISK-24455 - func_cdr: CDR_PROP leaks payload (Reported by
      Corey Farrell)
 * ASTERISK-24454 - app_queue: ao2_iterator not destroyed, causing
      leak (Reported by Corey Farrell)
 * ASTERISK-24430 - missing letter "p" in word response in
      OriginateResponse event documentation (Reported by Dafi Ni)
 * ASTERISK-24437 - Review implementation of ast_bridge_impart for
      leaks and document proper usage (Reported by Scott Griepentrog)
 * ASTERISK-24453 - manager: acl_change_sub leaks (Reported by
      Corey Farrell)
 * ASTERISK-24457 - res_fax: fax gateway frames leak (Reported by
      Corey Farrell)
 * ASTERISK-24458 - chan_phone fails to build on big endian systems
      (Reported by Tzafrir Cohen)
 * ASTERISK-21721 - SIP Failed to parse multiple Supported: headers
      (Reported by Olle Johansson)
 * ASTERISK-24304 - asterisk crashing randomly because of unistim
      channel (Reported by dhanapathy sathya)
 * ASTERISK-24190 - IMAP voicemail causes segfault (Reported by
      Nick Adams)
 * ASTERISK-24462 - res_pjsip: Stale qualify statistics after
      disablementation (Reported by Kevin Harwell)
 * ASTERISK-24465 - audiohooks list leaks reference to formats
      (Reported by Corey Farrell)
 * ASTERISK-24466 - app_queue: fix a couple leaks to struct
      call_queue (Reported by Corey Farrell)
 * ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled
      (Reported by Corey Farrell)
 * ASTERISK-24411 - [patch] Status of outbound registration is not
      changed upon unregistering. (Reported by John Bigelow)
 * ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream
      leaks (Reported by Corey Farrell)
 * ASTERISK-24480 - res_http_websockets: Module reference decrease
      below zero (Reported by Corey Farrell)
 * ASTERISK-24482 - func_talkdetect: Fix stasis message leak in
      audiohook callback (Reported by Corey Farrell)
 * ASTERISK-24487 - configuration: sections should be loadable as
      template even when not marked (Reported by Scott Griepentrog)
 * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
      unescapes semicolons ("\;" -> ";") turning them into comments
      (corruption) on rewrite of a config file (Reported by George
      Joseph)
 * ASTERISK-24438 - res_pjsip_multihomed.so blocks Asterisk reload
      when DNS settings invalid (Reported by Melissa Shepherd)
 * ASTERISK-24307 - Unintentional memory retention in stringfields
      (Reported by Etienne Lessard)
 * ASTERISK-24491 - Memory leak in res_hep (Reported by Zane
      Conkle)
 * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
      extra calls to ast_module_unref (Reported by Corey Farrell)
 * ASTERISK-24447 - Bridge DTMF hooks: Audio doesn't pass when
      waiting for more matching digits. (Reported by Richard Mudgett)
 * ASTERISK-24257 - agent must dial acceptdtmf twice to bridge to
      queue caller (Reported by Steve Pitts)
 * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
      (Reported by Corey Farrell)
 * ASTERISK-24250 - [patch] Voicemail with multi-recipients To:
      header fix (Reported by abelbeck)
 * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
      length exceeds 50 (roughly) national symbols (Reported by
      Dmitriy Bubnov)
 * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
      revision r227276 (Reported by Xavier Hienne)
 * ASTERISK-24505 - manager: http connections leak references
      (Reported by Corey Farrell)
 * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
      coverage are enabled (Reported by Corey Farrell)
 * ASTERISK-24444 - PBX: Crash when generating extension for
      pattern matching hint (Reported by Leandro Dardini)
 * ASTERISK-24489 - Crash: Asterisk crashes when converting RTCP
      packet to JSON for res_hep_rtcp and report blocks are greater
      than 1 (Reported by Gregory Malsack)
 * ASTERISK-24498 - Segmentation fault in res_hep_rtcp on attended
      transfer (Reported by Beppo Mazzucato)
 * ASTERISK-24501 - ARI: Moving a channel between bridges followed
      by a hangup can cause an ARI client to not receive an expected
      ChannelLeftBridge event before StasisEnd (Reported by Matt
      Jordan)
 * ASTERISK-24336 - PJSIP timer_min_se value under 90 causes crash
      (Reported by Leon Rowland)
 * ASTERISK-23651 - Reloading some modules that are loaded already,
      results in 'No such module' before a successful reload (Reported
      by Rusty Newton)
 * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
      endmarked users when last marked user leaves (Reported by Matt
      Jordan)
 * ASTERISK-15242 - transmit_refer leaks sip_refer structures
      (Reported by David Woolley)
 * ASTERISK-24508 - pjsip - REFER request from SNOM is rejected
      with "400 bad request" - DEBUG shows "Received a REFER without a
      parseable Refer-To" (Reported by Beppo Mazzucato)
 * ASTERISK-24535 - stringfields: Fix regression from fix for
      unintentional memory retention and another issue exposed by the
      fix (Reported by Corey Farrell)
 * ASTERISK-24471 - Crash - assert_fail in libc in
      pjmedia_sdp_neg_negotiateofrom /usr/local/lib/libpjmedia.so.2
      (Reported by yaron nahum)
 * ASTERISK-24528 - res_pjsip_refer: Sending INVITE with Replaces
      in-dialog with invalid target causes crash (Reported by Joshua
      Colp)
 * ASTERISK-24531 - res_pjsip_acl: ACLs not applied on initial
      module load (Reported by Matt Jordan)
 * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
      allow blocked addresses through (Reported by Matt Jordan)
 * ASTERISK-24542 - [patch]Failure showing codecs via 'core show
      channeltype <tech>' (Reported by snuffy)
 * ASTERISK-24533 - 2 threads created per chan_sip entry (Reported
      by xrobau)
 * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
      voicemail under high concurrency with an IMAP backend (Reported
      by David Duncan Ross Palmer)
 * ASTERISK-24572 - [patch]App_meetme is loaded without its
      defaults when the configuration file is missing (Reported by
      Nuno Borges)
 * ASTERISK-24573 - [patch]Out of sync conversation recording when
      divided in multiple recordings (Reported by NunowBorges)
 * ASTERISK-24537 - Stasis: StasisStart/StasisEnd events are not
      reliably transmitted during transfers (Reported by Matt Jordan)
 * ASTERISK-24556 - Asterisk 13 core dumps when calling from pjsip
      extension to another pjsip extension  (Reported by Abhay Gupta)

Improvements made in this release:
-----------------------------------
 * ASTERISK-24279 - Documentation: Clarify the behaviour of the CDR
      property 'unanswered' (Reported by Matt Jordan)
 * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
      column in the cel_odbc module (Reported by Etienne Lessard)
 * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
      recordings (Reported by Ben Smithurst)
 * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
      lookups (Reported by Birger "WIMPy" Harzenetter)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.1.0

Thank you for your continued support of Asterisk!

-----

The Asterisk Development Team has announced the release of Asterisk 13.2.0.

The release of Asterisk 13.2.0 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:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-24342 - PJSIP: Qualifying endpoints attempts to do them
      all at the same time. (Reported by Richard Mudgett)
 * ASTERISK-24514 - res_pjsip_outbound_registration: stack overflow
      when using non-default sorcery wizard (Reported by Kevin
      Harwell)
 * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
      from JSSIP (Reported by Badalian Vyacheslav)
 * ASTERISK-24607 - res_pjsip_session: re-INVITE with declined
      media streams results in 488 (Reported by Matt Jordan)
 * ASTERISK-24563 - Direct Media calls within private network
      sometimes get one way audio (Reported by Kevin Harwell)
 * ASTERISK-24604 - res_rtp_asterisk: Crash during restart due to
      race condition in accessing codec in stored ast_frame and codec
      core (Reported by Matt Jordan)
 * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
      enabled (Reported by Richard Mudgett)
 * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
      enabled (Reported by Andreas Steinmetz)
 * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
      casts char to unsigned int (Reported by Walter Doekes)
 * ASTERISK-24536 - AMI redirect with PJSIP fails to move extra
      channel (Reported by Niklas Larsson)
 * ASTERISK-24459 - bridge_native_rtp: Native RTP bridging is
      chosen for RTP compatible channels when the DTMF mode is not
      compatible (Reported by Yaniv Simhi)
 * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
      level - 'Remote address is null, most likely RTP has been
      stopped' (Reported by Rusty Newton)
 * ASTERISK-24513 - Local channel apparently leaked in off-nominal
      DTMF attended transfer (Reported by Mark Michelson)
 * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
      on startup (Reported by Richard Kenner)
 * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
      destination when 'sendrpid=yes' (in proxy environment) (Reported
      by Karsten Wemheuer)
 * ASTERISK-23841 - DTMF atxfer doesn't set CallerID for the recall
      calls to the transferrer. (Reported by Richard Mudgett)
 * ASTERISK-24376 - res_pjsip_refer: REFER request for remote
      session attempts to direct channel to external_replaces
      extension instead of context, without providing for the
      Referred-To SIP URI (Reported by Matt Jordan)
 * ASTERISK-24591 - Stasis() side of an ARI originated channel
      cannot be Redirected (Reported by Kinsey Moore)
 * ASTERISK-24049 - Asterisk Manager Interface: A number of list
      type responses aren't using astman_send_listack (Reported by
      Jonathan Rose)
 * ASTERISK-24637 - Channel re-enters Stasis() when it should not
      (Reported by John Bigelow)
 * ASTERISK-24474 - sip_to_pjsip.py lacks documentation and does
      not function (Reported by John Kiniston)
 * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
      (Reported by Kristian Høgh)
 * ASTERISK-20744 - [patch] Security event logging does not work
      over syslog (Reported by Michael Keuter)
 * ASTERISK-24665 - Configure check required for
      pjsip_get_dest_info() (Reported by Mark Michelson)
 * ASTERISK-23850 - Park Application does not respect Return
      Context Priority (Reported by Andrew Nagy)
 * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
      in the CFlags returned (Reported by Diederik de Groot)
 * ASTERISK-24655 - res_pjsip_outbound_publish: Hang on shutdown
      while attempting to publish (Reported by Kevin Harwell)
 * ASTERISK-24485 - res_pjsip cannot be unloaded or shutdown
      (Reported by Corey Farrell)
 * ASTERISK-24663 - [patch] Unnamed semaphore autoconf check fails
      on cross compilation (Reported by abelbeck)
 * ASTERISK-24624 - Transfer to invalid extension results in hung
      channel. (Reported by Zane Conkle)
 * ASTERISK-24615 - When Multiple Transports Exist in pjsip.conf,
      Incorrect External Addresses is Used in SIP Packets When
      Responding to INVITE (Reported by David Justl)
 * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
      voicemail is not deleted after review, hangup (Reported by LEI
      FU)
 * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
      32-bit packages on 64-bit hosts (Reported by Ben Klang)
 * ASTERISK-24600 - Stuck IAX channels, Asterisk stops responding
      to most traffic, potential deadlock (Reported by Jeff Collell)
 * ASTERISK-24560 - Creating a named ARI bridge twice causes a
      crash (Reported by Kinsey Moore)
 * ASTERISK-24682 - app_dial: Multiple DialEnd events emitted when
      MACRO_RESULT or GOSUB_RESULT are an unexpected value (Reported
      by Matt Jordan)
 * ASTERISK-24640 - Registration pending stays forever after sip
      reload (Reported by Max Man)
 * ASTERISK-24673 - outgoing sip registers cannot be removed or
      modified without doing restart (or doing module unload
      chan_sip.so) (Reported by Stefan Engström)
 * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
      m() option does not queue an MWI event (Reported by Gareth
      Palmer)
 * ASTERISK-24649 - Pushing of channel into bridge fails; Stasis
      fails to get app name (Reported by John Bigelow)
 * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
      column comparison for 'defaultuser' (Reported by
      HZMI8gkCvPpom0tM)
 * ASTERISK-24693 - Investigate and fix memory leaks in Asterisk
      (Reported by Kevin Harwell)
 * ASTERISK-24626 - Voicemail passwords not being stored in ARA
      (Reported by Paddy Grice)
 * ASTERISK-24539 - Compile fails on OSX because of sem_timedwait
      in bridge_channel.c (Reported by George Joseph)
 * ASTERISK-24544 - Compile fails on OSX Yosemite because of
      incorrect detection of htonll and ntohll (Reported by George
      Joseph)
 * ASTERISK-24723 - confbridge: CLI command 'confbridge list XXXX'
      no longer displays user menus (Reported by Matt Jordan)
 * ASTERISK-24721 - manager: ModuleLoad action incorrectly reports
      'module not found' during a Reload operation (Reported by Matt
      Jordan)
 * ASTERISK-24719 - ConfBridge recording channels get stuck when
      recording started/stopped more than once (Reported by Richard
      Mudgett)
 * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
      by Kevin Harwell)
 * ASTERISK-24728 - tcptls: Bad file descriptor error when
      reloading chan_sip (Reported by Kevin Harwell)
 * ASTERISK-24729 - Outbound registration not occuring on new
      registrations after reload. (Reported by Richard Mudgett)
 * ASTERISK-24676 - Security Vulnerability: URL request injection
      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
 * ASTERISK-24666 - Security Vulnerability: RTP not closed after
      sip call using unsupported codec (Reported by Y Ateya)
 * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
      versions (Reported by Jared Biel)
 * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
      Stephan Eisvogel)
 * ASTERISK-24736 - Memory Leak Fixes (Reported by Mark Michelson)
 * ASTERISK-24635 - PJSIP outbound PUBLISH crashes when no response
      is ever received (Reported by Marco Paland)
 * ASTERISK-24737 - When agent not logged in, agent status shows
      unavailable, queue status shows agent invalid (Reported by
      Richard Mudgett)

Improvements made in this release:
-----------------------------------
 * ASTERISK-24552 - ARI: Allow associating a channel as an
      initiator of an Origination for record keeping purposes
      (Reported by Matt Jordan)
 * ASTERISK-24553 - ARI/AMI: Include language in standard channel
      snapshot output (Reported by Matt Jordan)
 * ASTERISK-24643 - res_pjsip: Add user=phone option (Reported by
      Matt Jordan)
 * ASTERISK-24644 - res_pjsip_keepalive: Add keepalive module for
      connection-oriented transports. (Reported by Matt Jordan)
 * ASTERISK-24412 - [patch]Incomplete channel originate/continue
      handling with ARI (Reported by Nir Simionovich (GreenfieldTech -
      Israel))
 * ASTERISK-24678 - [PATCH] Added atxfer* settings to
      features.conf.sample (Reported by Niklas Larsson)
 * ASTERISK-24575 - [patch]Make capath work for res_pjsip (Reported
      by cloos)
 * ASTERISK-24671 - Missing docs for the CDR AMI Event (Reported by
      Dan Jenkins)
 * ASTERISK-24316 - For httpd server, need option to define server
      name for security purposes (Reported by Andrew Nagy)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.2.0

Thank you for your continued support of Asterisk!

-----

The Asterisk Development Team has announced the release of Asterisk 13.2.1.

The release of Asterisk 13.2.1 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:

* --- pjsip: resolve compatibility problem with ast_sip_session
  (Closes issue ASTERISK-24941. 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-13.2.1

Thank you for your continued support of Asterisk!

-----

The Asterisk Development Team has announced the release of Asterisk 13.3.0.

The release of Asterisk 13.3.0 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-24703 - ARI: Add the ability to "transfer" (redirect) a
      channel (Reported by Matt Jordan)
 * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
      (Reported by Dwayne Hubbard)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-24616 - Crash in res_format_attr_h264 due to invalid
      string copy (Reported by Yura Kocyuba)
 * ASTERISK-24748 - res_pjsip: If wizards explicitly configured in
      sorcery.conf false ERROR messages may occur (Reported by Joshua
      Colp)
 * ASTERISK-24769 - res_pjsip_sdp_rtp: Local ICE candidates leaked
      (Reported by Matt Jordan)
 * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
      res_odbc (Reported by ibercom)
 * ASTERISK-24479 - Enable REF_DEBUG for module references
      (Reported by Corey Farrell)
 * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
      fully disconnect underlying socket, leading to events being
      dropped with no additional information (Reported by Matt Jordan)
 * ASTERISK-24772 - ODBC error in realtime sippeers when device
      unregisters under MariaDB (Reported by Richard Miller)
 * ASTERISK-24752 - Crash in bridge_manager_service_req when bridge
      is destroyed by ARI during shutdown (Reported by Richard
      Mudgett)
 * ASTERISK-24741 - dtls_handler causes Asterisk to crash (Reported
      by Zane Conkle)
 * ASTERISK-24015 - app_transfer fails with PJSIP channels
      (Reported by Private Name)
 * ASTERISK-24727 - PJSIP: Crash experienced during multi-Asterisk
      transfer scenario. (Reported by Mark Michelson)
 * ASTERISK-24771 - ${CHANNEL(pjsip)} - segfault (Reported by
      Niklas Larsson)
 * ASTERISK-24716 - Improve pjsip log messages for presence
      subscription failure (Reported by Rusty Newton)
 * ASTERISK-24612 - res_pjsip: No information if a required sorcery
      wizard is not loaded (Reported by Joshua Colp)
 * ASTERISK-24768 - res_timing_pthread: file descriptor leak
      (Reported by Matthias Urlichs)
 * ASTERISK-24685 - "pjsip show version" CLI command (Reported by
      Joshua Colp)
 * ASTERISK-24632 - install_prereq script installs pjproject
      without IPv6 support (Reported by Rusty Newton)
 * ASTERISK-24085 - Documentation - We should remove or further
      document the 'contact' section in pjsip.conf (Reported by Rusty
      Newton)
 * ASTERISK-24791 - Crash in ast_rtcp_write_report (Reported by
      JoshE)
 * ASTERISK-24700 - CRASH: NULL channel is being passed to
      ast_bridge_transfer_attended() (Reported by Zane Conkle)
 * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
      (Reported by Corey Farrell)
 * ASTERISK-24799 - [patch] make fails with undefined reference to
      SSLv3_client_method (Reported by Alexander Traud)
 * ASTERISK-22670 - Asterisk crashes when processing ISDN AoC
      Events (Reported by klaus3000)
 * ASTERISK-24689 - Segfault on hangup after outgoing PRI-Euroisdn
      call (Reported by Marcel Manz)
 * ASTERISK-24740 - [patch]Segmentation fault on aoc-e event
      (Reported by Panos Gkikakis)
 * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
      for playing back messages stored in IMAP - play_message: No
      origtime (Reported by Graham Barnett)
 * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
      OSX with 64 bit integers (Reported by Corey Farrell)
 * ASTERISK-24796 - Codecs and bucket schema's prevent module
      unload (Reported by Corey Farrell)
 * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
      (Reported by Ashley Sanders)
 * ASTERISK-24499 - Need more explicit debug when PJSIP dialstring
      is invalid (Reported by Rusty Newton)
 * ASTERISK-24785 - 'Expires' header missing from 200 OK on
      REGISTER (Reported by Ross Beer)
 * ASTERISK-24677 - ARI GET variable on channel provides unhelpful
      response on non-existent variable (Reported by Joshua Colp)
 * ASTERISK-24797 - bridge_softmix: G.729 codec license held
      (Reported by Kevin Harwell)
 * ASTERISK-24812 - ARI: Creating channels through /channels
      resource always uses SLIN, which results in unneeded transcoding
      (Reported by Matt Jordan)
 * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
      thread ID being passed to pthread_kill (Reported by JoshE)
 * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
      fail (Reported by Terry Wilson)
 * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
      SRTP for audio, but they responded without it' is ambiguous and
      wrong in some cases (Reported by Rusty Newton)
 * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
      error response and BYE are sent to the caller (Reported by
      Makoto Dei)
 * ASTERISK-18105 - most of asterisk modules are unbuildable in
      cygwin environment (Reported by feyfre)
 * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
 * ASTERISK-24751 - Integer values in json payload to ARI cause
      asterisk to crash (Reported by jeffrey putnam)
 * ASTERISK-24838 - chan_sip: Locking inversion occurs when
      building a peer causes a peer poke during request handling
      (Reported by Richard Mudgett)
 * ASTERISK-24825 - Caller ID not recognized using
      Centrex/Distinctive dialing (Reported by Richard Mudgett)
 * ASTERISK-24830 - res_rtp_asterisk.c checks USE_PJPROJECT not
      HAVE_PJPROJECT (Reported by Stefan Engström)
 * ASTERISK-24840 - res_pjsip: conflicting endpoint identifiers
      (Reported by Kevin Harwell)
 * ASTERISK-24755 - Asterisk sends unexpected early BYE to
      transferrer during attended transfer when using a Stasis bridge
      (Reported by John Bigelow)
 * ASTERISK-24739 - [patch] - Out of files -- call fails --
      numerous files with inodes from under /usr/share/zoneinfo,
      mostly posixrules (Reported by Ed Hynan)
 * ASTERISK-23390 - NewExten Event with application AGI shows up
      before and after AGI runs (Reported by Benjamin Keith Ford)
 * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
      voicemail stored in LDAP (Reported by Graham Barnett)
 * ASTERISK-24808 - res_config_odbc: Improper escaping of
      backslashes occurs with MySQL (Reported by Javier Acosta)
 * ASTERISK-24807 - Missing mandatory field Max-Forwards (Reported
      by Anatoli)
 * ASTERISK-20850 - [patch]Nested functions aren't portable.
      Adapting RAII_VAR to use clang/llvm blocks to get the
      same/similar functionality. (Reported by Diederik de Groot)
 * ASTERISK-24872 - [patch] AMI PJSIPShowEndpoint closes AMI
      connection on error (Reported by Dmitriy Serov)
 * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
      by Frank DiGennaro)
 * ASTERISK-21038 - Bad command completion of "core set debug
      channel" (Reported by Richard Kenner)
 * ASTERISK-18708 - func_curl hangs channel under load (Reported by
      Dave Cabot)
 * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
      Atis Lezdins)
 * ASTERISK-24876 - Investigate reference leaks from
      tests/channels/local/local_optimize_away (Reported by Corey
      Farrell)
 * ASTERISK-24882 - chan_sip: Improve usage of REF_DEBUG (Reported
      by Corey Farrell)
 * ASTERISK-24817 - init_logger_chain: unreachable code block
      (Reported by Corey Farrell)
 * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
      snuffy)
 * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
      under OpenBSD (Reported by snuffy)

Improvements made in this release:
-----------------------------------
 * ASTERISK-24745 - [patch]Add no_answer to ARI hangup causes
      (Reported by Ben Merrills)
 * ASTERISK-24811 - asterisk-publication sorcery object does not
      use realtime (Reported by Matt Hoskins)
 * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
      Couldn't find mailbox %s in context (Reported by Graham Barnett)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.0

Thank you for your continued support of Asterisk!

-----

The Asterisk Development Team has announced the release of Asterisk 13.3.1.

The release of Asterisk 13.3.1 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:

* --- pjsip: resolve compatibility problem with ast_sip_seesion
  (Closes issue ASTERISK-24941. 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-13.3.1

Thank you for your continued support of Asterisk!

-----

The Asterisk Development Team has announced the release of Asterisk 13.4.0.

The release of Asterisk 13.4.0 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-24922 - ARI: Add the ability to intercept hold and
      raise an event (Reported by Matt Jordan)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-25112 - Logger: Configuration settings are not reset to
      default during reload. (Reported by Corey Farrell)
 * ASTERISK-24944 - main/audiohook.c change prevents G722 call
      recording (Reported by Ronald Raikes)
 * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
      or more digits (Reported by Makoto Dei)
 * ASTERISK-25086 - [patch]PJSIP crashes if endpoint missing in
      Dial() (Reported by snuffy)
 * ASTERISK-25089 - res_pjsip_config_wizard: Variable specified in
      templates aren't being processed correctly (Reported by George
      Joseph)
 * ASTERISK-25090 - CLI core show channel truncates cdr variables
      (Reported by snuffy)
 * ASTERISK-25085 - [patch]Potential crash after unload of
      func_periodic_hook or test_message (Reported by Corey Farrell)
 * ASTERISK-25083 - Message.c: Message channel becomes saturated
      with frames leading to spammy log messages (Reported by Jonathan
      Rose)
 * ASTERISK-25082 - Asterisk deletes message after doing a playback
      of an INBOX message using ast_vm_play when the Old folder is
      full for that mailbox. (Reported by Jonathan Rose)
 * ASTERISK-25041 - [patch]Broken column type checking in
      res_config_mysql addon (Reported by Alexandre Fournier)
 * ASTERISK-21893 - Segfault after call hangup, in
      ast_channel_hangupcause_set, at channel_internal_api.c (Reported
      by Alexandr Gordeev)
 * ASTERISK-25074 - Regression: Recent clang-related change broke
      cross compiling of Asterisk (Reported by Sebastian Kemper)
 * ASTERISK-25042 - asterisk.conf options override command-line
      options. (Reported by Corey Farrell)
 * ASTERISK-24442 - Outgoing call files don't work properly when
      set in the future (Reported by tootai)
 * ASTERISK-25057 - res_pjsip_pubsub: Crash in send_notify due to
      invalid root pointer in sub_tree (Reported by Matt Jordan)
 * ASTERISK-24938 - ARI Snoop Channel results in excessive
      escalating CPU usage (Reported by George Ladoff)
 * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
      ignore ISDN RESTART requests. (Reported by Richard Mudgett)
 * ASTERISK-25003 - Asterisk crashes on attended transfer (using
      feature) (Reported by Artem Volodin)
 * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
      contain waiting time (Reported by Etienne Lessard)
 * ASTERISK-25027 - Build System: Many ARI modules are missing
      dependencies. (Reported by Corey Farrell)
 * ASTERISK-25061 - pbx_config: Register manager actions with
      module version of macro. (Reported by Corey Farrell)
 * ASTERISK-25025 - Periodic crashes (in
      ast_channel_snapshot_create at stasis_channels.c) with Certified
      Asterisk 13. (Reported by Chet Stevens)
 * ASTERISK-25053 - Unit test category /main/presence missing
      trailing slash. (Reported by Corey Farrell)
 * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
      not respected, failover between DSNs doesn't work (Reported by
      JoshE)
 * ASTERISK-25054 - Formats interface's cannot be unregistered,
      needs to hold modules until shutdown. (Reported by Corey
      Farrell)
 * ASTERISK-24896 - [patch] Using force black background leads to
      colours not being reset (Reported by dant)
 * ASTERISK-25033 - Asterisk 13 (branch head) won't compile without
      PJSip (Reported by Peter Whisker)
 * ASTERISK-25028 - Build System: Unneeded defines in
      asterisk/buildopts.h (Reported by Corey Farrell)
 * ASTERISK-25048 - Astobj2: Initialization order wrong when both
      refdebug and AO2_DEBUG are both enabled. (Reported by Corey
      Farrell)
 * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
      cause code 44 after some time. (Reported by Denis Alberto
      Martinez)
 * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
      (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-25037 - res_pjsip_outbound_registration: Potential
      crash in off-nominal failure case when sending message (Reported
      by Joshua Colp)
 * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
      (Reported by Steve Davies)
 * ASTERISK-22790 - check_modem_rate() may return incorrect rate
      for V.27 (Reported by not here)
 * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
      to minrate=2400, then res_fax refuse to load (Reported by David
      Brillert)
 * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
      which is disallowed in res_fax's check_modem_rate (Reported by
      Matt Jordan)
 * ASTERISK-24996 - chan_pjsip: Creating Channel Causes Asterisk to
      Crash When Duplicate AOR Sections Exist in pjsip.conf (Reported
      by Ashley Sanders)
 * ASTERISK-25020 - Mismatched response to outgoing REGISTER
      request (Reported by Mark Michelson)
 * ASTERISK-25018 - pjsip show endpoints crashes asterisk when
      qualified aors present (Reported by Ivan Poddubny)
 * ASTERISK-24749 - ConfBridge: Wrong language on playing
      conf-hasjoin and conf-hasleft when played to bridge (Reported by
      Philippe Bolduc)
 * ASTERISK-24845 - pjsip send notify not working with Cisco phone
      (Reported by Carl Fortin)
 * ASTERISK-25004 - Crash in authenticated reinvite after
      originated T.38 FAX (Reported by Mark Michelson)
 * ASTERISK-24999 - PJSIP crashes with malformed contact line
      (Reported by snuffy)
 * ASTERISK-24998 - res_corosync:  res_corosync tries to load even
      if res_corosync.conf is missing (Reported by George Joseph)
 * ASTERISK-24997 - Astobj2: Some callers of __adjust_lock do not
      pre-check the object (Reported by Corey Farrell)
 * ASTERISK-24982 - res_pjsip_mwi: Unsolicited MWI NOTIFY only sent
      on mailbox changes (Reported by Joshua Colp)
 * ASTERISK-24991 - Check for ao2_alloc failure in
      __ast_channel_internal_alloc (Reported by Corey Farrell)
 * ASTERISK-24895 - After hangup on the side of the ISDN network no
      HangupRequest event comes for the dahdi channel. (Reported by
      Andrew Zherdin)
 * ASTERISK-24977 - Contacts that don't use qualify are being
      marked as unavailable (Reported by George Joseph)
 * ASTERISK-24774 - Segfault in ast_context_destroy with
      extensions.ael and extensions.conf (Reported by Corey Farrell)
 * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when
      channels have multiple native formats (Reported by Matt Jordan)
 * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
      to Fail (Reported by Ashley Sanders)
 * ASTERISK-24958 - Forwarding loop detection inhibits certain
      desirable scenarios (Reported by Mark Michelson)
 * ASTERISK-24863 - res_pjsip: No endpoint events raised via AMI
      when contacts cannot be reached/qualified (Reported by Dmitriy
      Serov)
 * ASTERISK-24869 - Asterisk segfaults on DAHDI attended transfer
      due to application (appl) being NULL on unbridged channel
      (Reported by viniciusfontes)
 * ASTERISK-24970 - Crash in res_pjsip_pubsub handling of failed
      notify (Reported by Scott Griepentrog)
 * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
      (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-24954 - Git migration: Asterisk version numbers are
      incompatible with the Test Suite (Reported by Matt Jordan)
 * ASTERISK-17608 - func_aes.so cannot be loaded if res_crypto /
      openssl not compiled (Reported by Warren Selby)
 * ASTERISK-24928 - [patch]t38_udptl_maxdatagram in pjsip.conf not
      honored (Reported by Juergen Spies)
 * ASTERISK-24835 - Early Media Not working with Chan SIP and
      Asterisk 13 (Reported by Andrew Nagy)
 * ASTERISK-21777 - Asterisk tries to transcode video instead of
      audio (Reported by Nick Ruggles)
 * ASTERISK-24380 - core: Native formats are set to h264 with
      certain audio/video codec configuration, resulting in path
      translation WARNINGs (Reported by Matt Jordan)
 * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
      into account (Reported by Frederic Van Espen)
 * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
      short (Reported by Y Ateya)
 * ASTERISK-24935 - res_pjsip_phoneprov_provider: Fix leaked
      OBJ_MULTIPLE iterator. (Reported by Corey Farrell)
 * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
      (Reported by Vadim)
 * ASTERISK-24933 - T38 fails negotiation (Reported by Jonathan
      Rose)
 * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
      byte prefix bug (Reported by Matt Jordan)
 * ASTERISK-21211 - chan_iax2 - unprotected access of
      iaxs[peer->callno] potentially results in segfault (Reported by
      Jaco Kroon)
 * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
      (Reported by Christoph Timm)
 * ASTERISK-24782 - StasisEnd event not present for channel that
      was swapped out for another after completing attended transfer
      (Reported by John Bigelow)
 * ASTERISK-24910 - "timer=no" and "timer=required" settings in
      pjsip.conf fail (Reported by Ray Crumrine)
 * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
      (Reported by Jeffrey C. Ollie)
 * ASTERISK-24914 - Division by zero in file.c when playback of
      voicemail with video as h264 (Reported by Marcello Ceschia)
 * ASTERISK-24899 - Parking fall-through behavior different in 13
      (Reported by Malcolm Davenport)
 * ASTERISK-24937 - [patch]res_pjsip_messaging: Messages may be
      sent out of order (Reported by Mark Michelson)
 * ASTERISK-24920 - Asterisk handles duplicate SIP requests as if
      they were each a new request (Reported by Mark Michelson)
 * ASTERISK-24857 - [patch] "timing test", pjsip incoming/outgoing
      calls, voicemail prompts and recordings all fail when using the
      kqueue timer source on FreeBSD 10.x (Reported by Justin T.
      Gibbs)
 * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
      detection in ast_malloc (Reported by Timo Teräs)
 * ASTERISK-24142 - CCSS: crash during shutdown due to device
      lookup in destroyed container (Reported by David Brillert)
 * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
      core restart now (Reported by Peter Katzmann)
 * ASTERISK-24805 - [patch] - ASAN: Race condition
      (heap-use-after-free) on asterisk closing (Reported by Badalian
      Vyacheslav)
 * ASTERISK-24881 - ast_register_atexit should only be used when
      absolutely needed (Reported by Corey Farrell)
 * ASTERISK-24731 - res_pjsip_session cannot be unloaded (Reported
      by Corey Farrell)
 * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
      (Reported by Kevin Harwell)
 * ASTERISK-14233 - [patch] Buddies are always auto-registered when
      processing the roster (Reported by Simon Arlott)
 * ASTERISK-24780 - [patch] - Buddies are always auto-registered
      when processing the roster (Reported by Simon Arlott)
 * ASTERISK-24781 - PJSIP: Unnecessary 180 Ringing messages sent
      with undesireabe consequences. (Reported by Richard Mudgett)

Improvements made in this release:
-----------------------------------
 * ASTERISK-25044 - sorcery:  Add ability to insert a new wizard
      into an object type's list (Reported by George Joseph)
 * ASTERISK-24892 - Super Awesome Company sound prompts (Reported
      by Rusty Newton)
 * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
      Hjelm)
 * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
      (Reported by Alexander Traud)
 * ASTERISK-25045 - vector:  Add new capabilities and unit tests
      (Reported by George Joseph)
 * ASTERISK-24706 - [patch]add auto-dtmf mode for pjsip (Reported
      by yaron nahum)
 * ASTERISK-25051 - Remove unneeded uses of optional_api providers.
      (Reported by Corey Farrell)
 * ASTERISK-25040 - pbx: Improve performance of reloads by making
      hint destruction more performant (Reported by Matt Jordan)
 * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
      Diederik de Groot)
 * ASTERISK-24949 - res_pjsip_outbound_registration: Backport line
      functionality (Reported by Joshua Colp)
 * ASTERISK-24965 - cel_pgsql - log_error string references CDR
      instead of CEL (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-24918 - pjsip: add CLI options to display global and
      system configuration (Reported by Scott Griepentrog)
 * ASTERISK-24862 - [patch] Support in-dialog OPTIONS (Reported by
      yaron nahum)
 * ASTERISK-24802 - stasis: set a channel variable on websocket
      disconnect error (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-13.4.0

Thank you for your continued support of Asterisk!

-----

The Asterisk Development Team has announced the release of Asterisk 13.5.0.

The release of Asterisk 13.5.0 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-25256 - [patch]Post AMI VarSet to empty string events
      when Asterisk deletes a dialplan variable. (Reported by Richard
      Mudgett)
 * ASTERISK-25067 - Sorcery Caching: Implement a new caching module
      (Reported by Matt Jordan)
 * ASTERISK-25040 - pbx: Improve performance of reloads by making
      hint destruction more performant (Reported by Matt Jordan)
 * ASTERISK-25114 - res_pjsip:  Add AMI etents for chan_pjsip
      contact lifecycle changes (Reported by George Joseph)
 * ASTERISK-25072 - res_pjsip_outbound_registration: line
      functionality. Additional check for using the request URI
      (Reported by Dmitriy Serov)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-25250 - chan_sip - Despite the channel being answered,
      caller on a call established via Local channel continues to hear
      ringback (Reported by Etienne Lessard)
 * ASTERISK-25253 - confbridge volume options and other volume
      controls such as func_volume don't work (Reported by Dmitriy
      Serov)
 * ASTERISK-25247 - choppy audio when spying on a g722 channel,
      chan_sip or chan_pjsip (Reported by hristo)
 * ASTERISK-24867 - Docs for 'e' option in ResetCDR say to use
      CDR_PROP instead, CDR_PROP docs are unclear (Reported by Rusty
      Newton)
 * ASTERISK-24853 - Documentation claims chan_sip outbound
      registrations support WS or WSS as valid transports (not true)
      (Reported by PSDK)
 * ASTERISK-25242 - PJSIP: No audio when Asterisk inside NAT and
      endpoints outside NAT - implement functionality similar to
      chan_sip 'rtpkeepalive'? (Reported by Mark Michelson)
 * ASTERISK-25258 - chan_pjsip: Incorrect format switch on received
      RTP packet (Reported by Joshua Colp)
 * ASTERISK-25257 - [patch]channels/sig_pri.h -> sig_pri_span ->
      force_restart_unavailable_chans in wrong scope (Reported by
      Patric Marschall)
 * ASTERISK-24934 - [patch]Asterisk manager output does not escape
      control characters (Reported by warren smith)
 * ASTERISK-25255 - Missing AMI VarSet events when setting to an
      empty string. (Reported by Richard Mudgett)
 * ASTERISK-25254 - Crash if dialplan sets ATTENDEDTRANSFER to an
      empty string before Park. (Reported by Richard Mudgett)
 * ASTERISK-25183 - PJSIP: Crash on NULL channel in
      chan_pjsip_incoming_response despite previous checks for NULL
      channel (Reported by Matt Jordan)
 * ASTERISK-25201 - Crash in PJSIP distributor on already free'd
      threadpool (Reported by Matt Jordan)
 * ASTERISK-24782 - StasisEnd event not present for channel that
      was swapped out for another after completing attended transfer
      (Reported by John Bigelow)
 * ASTERISK-25240 - bridge_native_rtp: Direct media wrongfully
      started when completing attended transfer (Reported by Joshua
      Colp)
 * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
      (Reported by Rusty Newton)
 * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
      BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
      (Reported by Dmitry Burilov)
 * ASTERISK-24550 - res_rtp_asterisk: Crash in
      ast_rtp_on_ice_complete during DTLS handshake (Reported by
      Osaulenko Alexander)
 * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
      Badalian Vyacheslav)
 * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
      by Stefan Engström)
 * ASTERISK-25127 - DTLS crashes following "Unable to cancel
      schedule ID" in dtls_srtp_check_pending (Reported by Dade
      Brandon)
 * ASTERISK-25168 - Random Core Dumps on Asterisk 13.4 PJSIP, in
      ast_channel_name at channel_internal_api.c (Reported by Carl
      Fortin)
 * ASTERISK-25115 - Crash related to func
      sip_resolve_invoke_user_callback of res_pjsip/pjsip_resolver.c
      (Reported by John Bigelow)
 * ASTERISK-25226 - chan_sip: Channel leak in branch 13 on early
      replaces call pickup (Reported by Walter Doekes)
 * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
      (Reported by Walter Doekes)
 * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
      in rtp_engine.c (Reported by Walter Doekes)
 * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
      (Reported by Walter Doekes)
 * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
      Bad file descriptor" (Reported by Barry Chern)
 * ASTERISK-25165 - Testsuite - Sorcery memory cache leaks
      (Reported by Corey Farrell)
 * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
      13.4 (Reported by cervajs)
 * ASTERISK-25196 - res_pjsip_nat: rewrite_contact should not be
      applied to Contact header when Record-Route headers are present
      (Reported by Mark Michelson)
 * ASTERISK-24907 - res_pjsip_outbound_registration: crash during
      unload if registration attempts are still occuring (Reported by
      Kevin Harwell)
 * ASTERISK-25204 - res_pjsip_refer: Duplicated Referred-By or
      Replaces headers on outbound INVITEs. (Reported by Mark
      Michelson)
 * ASTERISK-25171 - Early completion of feature code attended
      transfer results in intermittent one-way audio, "ghost ringing"
      and robotic sound. (Reported by Rusty Newton)
 * ASTERISK-25189 - AMI: Add Linkedid header to standard channel
      snapshot information. (Reported by Richard Mudgett)
 * ASTERISK-25172 - Crash in channels/sip/sip blind
      transfer/caller_refer_only test in
      ast_format_cap_append_from_cap during ast_request (Reported by
      Matt Jordan)
 * ASTERISK-25180 - res_pjsip_mwi: Unsolicited MWI requires reload
      (Reported by Joshua Colp)
 * ASTERISK-25182 - [patch] on CLI sip reload, new codecs get
      appended only (Reported by Alexander Traud)
 * ASTERISK-25163 - Deadlock in chan_sip between reload of sip peer
      container and MWI Stasis callback (Reported by Dmitriy Serov)
 * ASTERISK-25091 - Asterisk REST API - bridge.addChannel crash
      asterisk when calling channel hangup while adding to bridge
      (Reported by Ilya Trikoz)
 * ASTERISK-24900 - Manager event ParkedCallSwap is not documented
      (Reported by Rusty Newton)
 * ASTERISK-25162 - func_pjsip_aor: Leak of contact in iterator
      (Reported by Corey Farrell)
 * ASTERISK-25158 - res_pjsip: Add option to use AAL2 packing when
      negotiating g.726 (Reported by Kevin Harwell)
 * ASTERISK-24344 - CDR_PROP(disable) disables CDR only for first
      dialed party (Reported by Janusz Karolak)
 * ASTERISK-24443 - CDR fields (dst, dcontext) empty in transfer
      call started from Macro (Reported by Arveno Santoro)
 * ASTERISK-25154 - [patch]fromtag may need to be updatep after
      successful call dialog match (Reported by Damian Ivereigh)
 * ASTERISK-25156 - chan_pjsip CHAN_START cel event lacks the
      correct context and exten (Reported by cloos)
 * ASTERISK-25157 - bridging: Performing a blonde transfer does not
      result in connected line updates (Reported by Joshua Colp)
 * ASTERISK-25087 - Asterisk segfault when using Directory
      application with alias option and specific mailbox configuration
      (Reported by Chet Stevens)
 * ASTERISK-24983 - IAX deadlock between hangup and scheduled
      actions (ex. largrq) (Reported by Y Ateya)
 * ASTERISK-25096 - [patch]Segfault when registering over
      websockets with PJSIP (in ast_sockaddr_isnull at
      /include/asterisk/netsock2.h) (Reported by Josh Kitchens)
 * ASTERISK-24963 - ASAN: heap-use-after-free with PJSIP and WSS
      (Reported by Badalian Vyacheslav)
 * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
      but asterisk doesn't detect it. (Reported by ibercom)
 * ASTERISK-25094 - PBX core: Investigate thread safety issues
      (Reported by Corey Farrell)
 * ASTERISK-25148 - res_pjsip NULL channel audit (Reported by Mark
      Michelson)
 * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
      | adpcm | ipc10} (Reported by Badalian Vyacheslav)
 * ASTERISK-25137 - endpoint stasis messages are delivered twice
      (Reported by Vitezslav Novy)
 * ASTERISK-25116 - res_pjsip:  Two PeerStatus AMI messages are
      sent for every status change (Reported by George Joseph)
 * ASTERISK-25131 - chan_pjsip: In-dialog authentication not
      handled. (Reported by Richard Mudgett)
 * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
      that end with ::80 (Reported by Mark Petersen)
 * ASTERISK-25122 - Large SIP packet received via pjsip over
      websocket crashes Asterisk  (Reported by Ivan Poddubny)
 * ASTERISK-25121 - Stasis: Fix unsafe use of stasis_unsubscribe in
      modules. (Reported by Corey Farrell)
 * ASTERISK-24988 - func_talkdetect: Test is bouncing sporadically
      (Reported by Joshua Colp)
 * ASTERISK-25105 - res_pjsip:  Possible incompatibility between
      qualify_timeout and pjproject-2.4 (Reported by George Joseph)
 * ASTERISK-25117 - res_mwi_external_ami: Fix manager action
      registrations. (Reported by Corey Farrell)

New Features made in this release:
-----------------------------------
 * ASTERISK-25259 - chan_pjsip: Add rtptimeout support (Reported by
      Joshua Colp)
 * ASTERISK-25238 - ARI: Support push configuration (Reported by
      Matt Jordan)
 * ASTERISK-25173 - ARI: Add the ability to load/reload/unload an
      Asterisk module (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-13.5.0

Thank you for your continued support of Asterisk!

-----

The Asterisk Development Team has announced the release of Asterisk 13.6.0.

The release of Asterisk 13.6.0 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-25377 - res_pjsip: Change default "From user" from UUID
      to something more palatable (Reported by Mark Michelson)
 * ASTERISK-25252 - ARI: Add the ability to manipulate log channels
      (Reported by Matt Jordan)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-25449 - main/sched: Regression introduced by
      5c713fdf18f causes erroneous duplicate RTCP messages; other
      potential scheduling issues in chan_sip/chan_skinny (Reported by
      Matt Jordan)
 * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
      ICE is not enabled (Reported by Joshua Colp)
 * ASTERISK-25383 - Core dumps on startup and shutdown with
      MALLOC_DEBUG enabled (Reported by yaron nahum)
 * ASTERISK-25423 - Caller gets no Connected line update during
      call pickup. (Reported by Richard Mudgett)
 * ASTERISK-25305 - Dynamic logger channels can be added multiple
      times (Reported by Mark Michelson)
 * ASTERISK-25418 - On-hold channels redirected out of a bridge
      appear to still be on hold (Reported by Mark Michelson)
 * ASTERISK-25384 - Regular Asterisk crashes when using Page
      application. "user_data is NULL" (Reported by Chet Stevens)
 * ASTERISK-25407 - Asterisk fails to log to multiple syslog
      destinations (Reported by Elazar Broad)
 * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
      populated (Reported by Kevin Harwell)
 * ASTERISK-25394 - pbx: Incorrect device and presence state when
      changing hint details (Reported by Joshua Colp)
 * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
      invalid SIP (Reported by Walter Doekes)
 * ASTERISK-25399 - app_queue: AgentComplete event has wrong reason
      (Reported by Kevin Harwell)
 * ASTERISK-25185 - Segfault in app_queue on transfer scenarios
      (Reported by Etienne Lessard)
 * ASTERISK-25353 - [patch] Transcoding while different in Frame
      size = Frames lost (Reported by Alexander Traud)
 * ASTERISK-25325 - ARI PUT reload chan_sip HTTP response 404
      (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-25390 - default_from_user can crash with certain
      configuration backends (Reported by Mark Michelson)
 * ASTERISK-25387 - res_pjsip_nat: Malformed REGISTER request
      causes NAT'd Contact header to not be rewritten (Reported by
      Matt Jordan)
 * ASTERISK-25227 - No audio at in-band announcements in ooh323
      channel (Reported by Alexandr Dranchuk)
 * ASTERISK-25369 - res_parking: ParkAndAnnounce - Inheritable
      variables aren't applied to the announcer channel (Reported by
      Jonathan Rose)
 * ASTERISK-25295 - res_pjsip crash - pjsip_uri_get_uri at
      /usr/include/pjsip/sip_uri.h (Reported by Dmitriy Serov)
 * ASTERISK-25381 - res_pjsip: AoRs deleted via ARI (or other
      mechanism) do not destroy their related contacts (Reported by
      Matt Jordan)
 * ASTERISK-25367 - pbx: Long pattern match hints may cause "core
      show hints" to crash (Reported by Joshua Colp)
 * ASTERISK-25365 - Persistent subscriptions have extra
      Content-Length/corrupted messages (Reported by Mark Michelson)
 * ASTERISK-25362 - Deadlock due to presence state callback
      (Reported by Mark Michelson)
 * ASTERISK-25356 - res_pjsip_sdp_rtp: Multiple keepalive scheduled
      items may exist (Reported by Joshua Colp)
 * ASTERISK-25355 - sched: ast_sched_del may return prematurely due
      to spurious wakeup (Reported by Joshua Colp)
 * ASTERISK-25318 -
      tests/rest_api/applications/subscribe-endpoint/nominal/resource:
      Sporadically failing (Reported by Joshua Colp)
 * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
      cause on call pickup (Reported by Joshua Colp)
 * ASTERISK-25342 - res_pjsip: Repeated usage of pj_gethostip may
      block (Reported by Joshua Colp)
 * ASTERISK-25341 - bridge: Hangups may get lost when executing
      actions (Reported by Joshua Colp)
 * ASTERISK-25339 - res_pjsip: Empty "auth" sections from
      non-config backgrounds are interpreted as valid (Reported by
      Matt Jordan)
 * ASTERISK-25215 - Differences in queue.log between Set
      QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
      Gaetz)
 * ASTERISK-25322 - Crash occurs when using MixMonitor with t() or
      r() options. (Reported by Richard Mudgett)
 * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
      for wrong or non existent peer on invite (Reported by Kevin
      Harwell)
 * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
      tones. (Reported by Richard Mudgett)
 * ASTERISK-25312 - res_http_websocket: Terminate connection on
      fatal cases (Reported by Joshua Colp)
 * ASTERISK-25306 - Persistent subscriptions can save multiple SIP
      messages at once, leading to potential crashes. (Reported by
      Mark Michelson)
 * ASTERISK-25309 - [patch] iLBC 20 advertised (Reported by
      Alexander Traud)
 * ASTERISK-25304 - res_pjsip: XML sanitization may write past
      buffer (Reported by Joshua Colp)
 * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
      Firefox 39 - add ECDH support and fallback to prime256v1
      (Reported by Stefan Engström)
 * ASTERISK-25296 - RTP performance issue with several channel
      drivers. (Reported by Richard Mudgett)
 * ASTERISK-25297 - Crashes running
      channels/pjsip/resolver/srv/failover/in_dialog testsuite tests
      (Reported by Richard Mudgett)
 * ASTERISK-25292 - Testuite:
      tests/apps/bridge/bridge_wait/bridge_wait_e_options fails
      (Reported by Kevin Harwell)
 * ASTERISK-25271 - Parking & blind transfer: Transferer channel
      not hung up if no MOH (Reported by Kevin Harwell)

Improvements made in this release:
-----------------------------------
 * ASTERISK-24870 - ARI: Subscriptions to bridges generally not
      super useful (Reported by Matt Jordan)
 * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
      defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.6.0

Thank you for your continued support of Asterisk!

Revision 1.1.1.1 / (download) - annotate - [select for diffs] (vendor branch), Sat Dec 5 23:29:05 2015 UTC (3 years, 5 months ago) by jnemeth
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Changes since 1.1: +0 -0 lines
Diff to previous 1.1 (colored)

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Branch: MAIN

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