[asterisk-dev] Asterisk 13.7.0-rc1 Now Available

Asterisk Development Team asteriskteam at digium.com
Tue Dec 15 14:10:21 CST 2015 

The Asterisk Development Team has announced the first release candidate of
Asterisk 13.7.0. This release candidate is available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.7.0-rc1 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 candidate:

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-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 Nemčić)
 * 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 candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.7.0-rc1

Thank you for your continued support of Asterisk!

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