[asterisk-dev] Asterisk 11.14.0-rc1 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the first release candidate of
Asterisk 11.14.0. This release candidate is available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.14.0-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-24348 - Built-in editline tab complete segfault with
      MALLOC_DEBUG (Reported by Walter Doekes)
 * ASTERISK-24335 - [PATCH] Asterisk incorrectly responds 503 to
      INVITE retransmissions of rejected calls (Reported by Torrey
      Searle)
 * ASTERISK-23768 - [patch] Asterisk man page contains a (new)
      unquoted minus sign (Reported by Jeremy Lainé)
 * ASTERISK-24357 - [fax] Out of bounds error in update_modem_bits
      (Reported by Jeremy Lainé)
 * ASTERISK-20567 - bashism in autosupport (Reported by Tzafrir
      Cohen)
 * ASTERISK-22945 - [patch] Memory leaks in chan_sip.c with
      realtime peers (Reported by ibercom)
 * ASTERISK-24384 - chan_motif: format capabilities leak on module
      load error (Reported by Corey Farrell)
 * ASTERISK-24385 - chan_sip: process_sdp leaks on an error path
      (Reported by Corey Farrell)
 * ASTERISK-24378 - Release AMI connections on shutdown (Reported
      by Corey Farrell)
 * ASTERISK-24354 - AMI sendMessage closes AMI connection on error
      (Reported by Peter Katzmann)
 * ASTERISK-24390 - astobj2: REF_DEBUG reports false leaks with
      ao2_callback with OBJ_MULTIPLE (Reported by Corey Farrell)
 * ASTERISK-24326 - res_rtp_asterisk: ICE-TCP candidates are
      incorrectly attempted (Reported by Joshua Colp)
 * ASTERISK-24011 - [patch]safe_asterisk tries to set ulimit -n too
      high on linux systems with lots of RAM (Reported by Michael
      Myles)
 * ASTERISK-24383 - res_rtp_asterisk: Crash if no candidates
      received for component (Reported by Kevin Harwell)
 * ASTERISK-20784 - Failure to receive an ACK to a SIP Re-INVITE
      results in a SIP channel leak (Reported by NITESH BANSAL)
 * ASTERISK-15879 - [patch] Failure to receive an ACK to a SIP
      Re-INVITE results in a SIP channel leak (Reported by Torrey
      Searle)
 * ASTERISK-24406 - Some caller ID strings are parsed differently
      since 11.13.0 (Reported by Etienne Lessard)
 * ASTERISK-24325 - res_calendar_ews: cannot be used with neon 0.30
      (Reported by Tzafrir Cohen)
 * ASTERISK-13797 - [patch] relax badshell tilde test (Reported by
      Tzafrir Cohen)
 * ASTERISK-22791 - asterisk sends Re-INVITE after receiving a BYE
      (Reported by Paolo Compagnini)
 * ASTERISK-18923 - res_fax_spandsp usage counter is wrong
      (Reported by Grigoriy Puzankin)
 * ASTERISK-24392 - res_fax: fax gateway sessions leak (Reported by
      Corey Farrell)
 * ASTERISK-24393 - rtptimeout=0 doesn't disable rtptimeout
      (Reported by Dmitry Melekhov)
 * ASTERISK-23846 - Unistim multilines. Loss of voice after second
      call drops (on a second line). (Reported by Rustam Khankishyiev)
 * ASTERISK-24063 - [patch]Asterisk does not respect outbound proxy
      when sending qualify requests (Reported by Damian Ivereigh)
 * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of
      SSLv3, security fix POODLE (CVE-2014-3566) (Reported by
      abelbeck)
 * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling
      against libsrtp-1.5.0 (Reported by Patrick Laimbock)
 * 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-24457 - res_fax: fax gateway frames leak (Reported by
      Corey Farrell)
 * 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-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-24476 - main/app.c / app_voicemail: ast_writestream
      leaks (Reported by Corey Farrell)

For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.14.0-rc1

Thank you for your continued support of Asterisk!