[asterisk-dev] Asterisk 1.8.27.0-rc1 Now Available

Asterisk Development Team asteriskteam at digium.com
Fri Mar 28 15:12:45 CDT 2014


The Asterisk Development Team has announced the first release candidate of
Asterisk 1.8.27.0. This release candidate is available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 1.8.27.0-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-22790 - check_modem_rate() may return incorrect rate
      for V.27 (Reported by Paolo Compagnini)
 * ASTERISK-23061 - [Patch] 'textsupport' setting not mentioned in
      sip.conf.sample (Reported by Eugene)
 * ASTERISK-23028 - [patch] Asterisk man pages contains unquoted
      minus signs (Reported by Jeremy Lainé)
 * ASTERISK-23046 - Custom CDR fields set during a GoSUB called
      from app_queue are not inserted (Reported by Denis Pantsyrev)
 * ASTERISK-23027 - [patch] Spelling typo "transfered" instead of
      "transferred" (Reported by Jeremy Lainé)
 * ASTERISK-23008 - Local channels loose CALLERID name when DAHDI
      channel connects (Reported by Michael Cargile)
 * ASTERISK-23100 - [patch] In chan_mgcp the ident in transmitted
      request and request queue may differ - fix for locking (Reported
      by adomjan)
 * ASTERISK-22988 - [patch]T38 , SIP 488 after Rejecting image
      media offer due to invalid or unsupported syntax (Reported by
      adomjan)
 * ASTERISK-22861 - [patch]Specifying a null time as parameter to
      GotoIfTime or ExecIfTime causes segmentation fault (Reported by
      Sebastian Murray-Roberts)
 * ASTERISK-17837 - extconfig.conf - Maximum Include level (1)
      exceeded (Reported by pz)
 * ASTERISK-22662 - Documentation fix? - queues.conf says
      persistentmembers defaults to yes, it appears to lie (Reported
      by Rusty Newton)
 * ASTERISK-23134 - [patch] res_rtp_asterisk port selection cannot
      handle selinux port restrictions (Reported by Corey Farrell)
 * ASTERISK-23220 - STACK_PEEK function with no arguments causes
      crash/core dump (Reported by James Sharp)
 * ASTERISK-19773 - Asterisk crash on issuing Asterisk-CLI 'reload'
      command multiple times on cli_aliases (Reported by Joel Vandal)
 * ASTERISK-22757 - segfault in res_clialiases.so on reload when
      mapping "module reload" command (Reported by Gareth Blades)
 * ASTERISK-17727 - [patch] TLS doesn't get all certificate chain
      (Reported by LN)
 * ASTERISK-23178 - devicestate.h: device state setting functions
      are documented with the wrong return values (Reported by
      Jonathan Rose)
 * ASTERISK-23297 - Asterisk 12, pbx_config.so segfaults if
      res_parking.so is not loaded, or if res_parking.conf has no
      configuration (Reported by CJ Oster)
 * ASTERISK-23069 - Custom CDR variable not recorded when set in
      macro called from app_queue (Reported by Bryan Anderson)
 * ASTERISK-19499 - ConfBridge MOH is not working for transferee
      after attended transfer (Reported by Timo Teräs)
 * ASTERISK-23261 - [patch]Output mixup in
      ${CHANNEL(rtpqos,audio,all)} (Reported by rsw686)
 * ASTERISK-23260 - [patch]ForkCDR v option does not keep CDR
      variables for subsequent records (Reported by zvision)
 * ASTERISK-23141 - Asterisk crashes on Dial(), in
      pbx_find_extension at pbx.c (Reported by Maxim)
 * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
      to minrate=2400, then res_fax refuse to load (Reported by David
      Brillert)
 * ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set
      - probably introduced in 11.7.0 (Reported by OK)
 * ASTERISK-23323 - [patch]chan_sip: missing p->owner checks in
      handle_response_invite (Reported by Walter Doekes)
 * ASTERISK-23382 - [patch]Build System: make -qp can corrupt
      menuselect-tree and related files (Reported by Corey Farrell)
 * ASTERISK-23406 - [patch]Fix typo in "sip show peer" (Reported by
      ibercom)
 * ASTERISK-23310 - bridged channel crashes in bridge_p2p_rtp_write
      (Reported by Jeremy Lainé)
 * ASTERISK-23104 - Specifying the SetVar AMI without a Channel
      cause Asterisk to crash (Reported by Joel Vandal)
 * ASTERISK-23383 - Wrong sense test on stat return code causes
      unchanged config check to break with include files. (Reported by
      David Woolley)
 * ASTERISK-17523 - Qualify for static realtime peers does not work
      (Reported by Maciej Krajewski)
 * ASTERISK-21406 - [patch] chan_sip deadlock on monlock between
      unload_module and do_monitor (Reported by Corey Farrell)
 * ASTERISK-23373 - [patch]Security: Open FD exhaustion with
      chan_sip Session-Timers (Reported by Corey Farrell)
 * ASTERISK-23340 - Security Vulnerability: stack allocation of
      cookie headers in loop allows for unauthenticated remote denial
      of service attack (Reported by Matt Jordan)
 * ASTERISK-23488 - Logic error in callerid checksum processing
      (Reported by Russ Meyerriecks)
 * ASTERISK-20841 - fromdomain not honored on outbound INVITE
      request (Reported by Kelly Goedert)
 * ASTERISK-22079 - Segfault: INTERNAL_OBJ (user_data=0x6374652f)
      at astobj2.c:120 (Reported by Jamuel Starkey)
 * ASTERISK-23509 - [patch]SayNumber for Polish language tries to
      play empty files for numbers divisible by 100 (Reported by
      zvision)
 * ASTERISK-23391 - Audit dialplan function usage of channel
      variable (Reported by Corey Farrell)
 * ASTERISK-23548 - POST to ARI sometimes returns no body on
      success (Reported by Scott Griepentrog)

Improvements made in this release:
-----------------------------------
 * ASTERISK-22980 - [patch]Allow building cdr_radius and cel_radius
      against libfreeradius-client (Reported by Jeremy Lainé)
 * ASTERISK-22661 - Unable to exit ChanSpy if spied channel does
      not have a call in progress (Reported by Chris Hillman)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.27.0-rc1

Thank you for your continued support of Asterisk!



More information about the asterisk-dev mailing list