asterisk-lupaEl equipo de desarrolladores de Asterisk acaba de publicar la nueva versión Asterisk 13.3.0

Puedes ver la lista de cambios de esta versión en el siguiente enlace:
Descargar Changelog

También puedes descargar esta versión en el siguiente enlace:
Descargar Asterisk

El anuncio oficial:

The Asterisk Development Team has announced the release of Asterisk 13.3.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

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!

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