[asterisk-dev] Asterisk 1.8.32.0-rc1 Now Available

Asterisk Development Team asteriskteam at digium.com
Mon Nov 3 16:54:39 CST 2014


The Asterisk Development Team has announced the first release candidate of
Asterisk 1.8.32.0. This release candidate is available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 1.8.32.0-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-22945 - [patch] Memory leaks in chan_sip.c with
      realtime peers (Reported by ibercom)
 * ASTERISK-24390 - astobj2: REF_DEBUG reports false leaks with
      ao2_callback with OBJ_MULTIPLE (Reported by Corey Farrell)
 * ASTERISK-24011 - [patch]safe_asterisk tries to set ulimit -n too
      high on linux systems with lots of RAM (Reported by Michael
      Myles)
 * 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-24393 - rtptimeout=0 doesn't disable rtptimeout
      (Reported by Dmitry Melekhov)
 * 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-21721 - SIP Failed to parse multiple Supported: headers
      (Reported by Olle Johansson)
 * ASTERISK-24190 - IMAP voicemail causes segfault (Reported by
      Nick Adams)
 * 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-1.8.32.0-rc1

Thank you for your continued support of Asterisk!



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