[asterisk-users] Asterisk 13.14.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the release of Asterisk 13.14.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

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!