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’s 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!

4 files changed, 4 insertions, 4 deletions

diff --git a/comms/asterisk13/Makefile b/comms/asterisk13/Makefile
index ab0f75bc93d..abbba7bb96f 100644
--- a/comms/asterisk13/Makefile
+++ b/comms/asterisk13/Makefile
@@ -1,4 +1,4 @@
-# $NetBSD: Makefile,v 1.1.1.1 2015/12/05 23:29:05 jnemeth Exp $
+# $NetBSD: Makefile,v 1.2 2015/12/05 23:42:44 jnemeth Exp $
 #
 # NOTE: when updating this package, there are two places that sound
 #       tarballs need to be checked
diff --git a/comms/asterisk13/PLIST b/comms/asterisk13/PLIST
index 86254ae007f..c54c0ac5e7b 100644
--- a/comms/asterisk13/PLIST
+++ b/comms/asterisk13/PLIST
@@ -1,4 +1,4 @@
-@comment $NetBSD: PLIST,v 1.1.1.1 2015/12/05 23:29:08 jnemeth Exp $
+@comment $NetBSD: PLIST,v 1.2 2015/12/05 23:42:44 jnemeth Exp $
 include/asterisk.h
 include/asterisk/_private.h
 include/asterisk/abstract_jb.h
diff --git a/comms/asterisk13/distinfo b/comms/asterisk13/distinfo
index c91ace0c54d..9657b8f6401 100644
--- a/comms/asterisk13/distinfo
+++ b/comms/asterisk13/distinfo
@@ -1,4 +1,4 @@
-$NetBSD: distinfo,v 1.1.1.1 2015/12/05 23:29:08 jnemeth Exp $
+$NetBSD: distinfo,v 1.2 2015/12/05 23:42:44 jnemeth Exp $
 
 SHA1 (asterisk-13.6.0/asterisk-13.6.0.tar.gz) = 320d3b83f6d4641db76c957738b21a7b8eec938b
 RMD160 (asterisk-13.6.0/asterisk-13.6.0.tar.gz) = c6cdb801c65f3539d16c6db1e10bd75e6807684e
diff --git a/comms/asterisk13/options.mk b/comms/asterisk13/options.mk
index 2078962fd31..f72d51ad596 100644
--- a/comms/asterisk13/options.mk
+++ b/comms/asterisk13/options.mk
@@ -1,4 +1,4 @@
-# $NetBSD: options.mk,v 1.1.1.1 2015/12/05 23:29:09 jnemeth Exp $
+# $NetBSD: options.mk,v 1.2 2015/12/05 23:42:44 jnemeth Exp $
 
 PKG_OPTIONS_VAR=		PKG_OPTIONS.asterisk
 PKG_SUPPORTED_OPTIONS=		zaptel x11 unixodbc ilbc webvmail ldap spandsp