[asterisk-dev] Asterisk 11.21.0-rc1 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the first release candidate of
Asterisk 11.21.0. This release candidate is available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.21.0-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:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
      thread of asterisk is not released (Reported by Hiroaki Komatsu)
 * 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-25498 - Asterisk crashes when negotiating g729 without
      that module installed (Reported by Ben Langfeld)
 * ASTERISK-25476 - chan_sip loses registrations after a while
      (Reported by Michael Keuter)
 * 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-25552 - hashtab: Improve NULL tolerance (Reported by
      Joshua Colp)
 * 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-25537 - [patch] format-attribute module: RFC or
      internal defaults? (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-25434 - Compiler flags not reported in 'core show
      settings' despite usage during compilation (Reported by Rusty
      Newton)
 * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
      bounds and missing paren issues (Reported by George Joseph)
 * ASTERISK-7803 - [patch] Update the maximum packetization values
      in frame.c (Reported by dea)
 * 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-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-25391 - AMI GetConfigJSON returns invalid JSON
      (Reported by Bojan Nemčić)
 * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
      ICE is not enabled (Reported by Joshua Colp)

Improvements made in this release:
-----------------------------------
 * 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-11.21.0-rc1

Thank you for your continued support of Asterisk!