asterisk announce - Sep 2014 - Asterisk 12.6.0 Now Available

The Asterisk Development Team has announced the release of Asterisk 12.6.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 12.6.0 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:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-24027 - MixMonitor AMI action called during AGI
      execution from bridge feature causes channel to leave AGI has
      hung up (Reported by Matt Jordan)
 * ASTERISK-24236 - res_hep_rtcp: Module incorrectly depends on
      pjsip (Reported by Matt Jordan)
 * ASTERISK-24032 - Gentoo compilation emits warning:
      "_FORTIFY_SOURCE" redefined (Reported by Kilburn)
 * ASTERISK-24225 - Dial option z is broken (Reported by
      dimitripietro)
 * ASTERISK-24234 - app_meetme: Crash on conference shutdown due to
      NULL channel passed to meetme_stasis_generate_msg() (Reported by
      Shaun Ruffell)
 * ASTERISK-24043 - ARI /continue fails to actually continue into
      the dialplan (Reported by Krandon Bruse)
 * ASTERISK-24245 - gcc 4.1.2 complains of files that do not end
      with newlines (Reported by Shaun Ruffell)
 * ASTERISK-24229 - ARI: playback of sounds implicitly answers
      channel, preventing early media playback (Reported by Matt
      Jordan)
 * ASTERISK-24178 - [patch]fromdomainport used even if not set
      (Reported by Elazar Broad)
 * ASTERISK-22252 - res_musiconhold cleanup - REF_DEBUG reload
      warnings and ref leaks (Reported by Walter Doekes)
 * ASTERISK-23994 - res_pjsip_sdp_rtp: owner address in SDP may not
      be fully qualified domainname (Reported by Private Name)
 * ASTERISK-24147 - ARI: channel hangup crashes asterisk process
      (Reported by Edvin Vidmar)
 * ASTERISK-23997 - chan_sip: port incorrectly incremented for RTCP
      ICE candidates in SDP answer (Reported by Badalian Vyacheslav)
 * ASTERISK-24143 - pjsip: Outbound call to WebRTC UA fails to
      transmit ACK on received 200 OK (Reported by Aleksei Kulakov)
 * ASTERISK-24019 - When a Music On Hold stream starts it restarts
      at beginning of file. (Reported by Jason Richards)
 * ASTERISK-23767 - [patch] Dynamic IAX2 registration stops trying
      if ever not able to resolve (Reported by David Herselman)
 * ASTERISK-24264 - ARI: Adding a channel to a holding bridge
      automatically starts MOH (Reported by Samuel Galarneau)
 * ASTERISK-24212 - testsuite: Sporadic crash due to assert on
      stopping RTP engine (Reported by Matt Jordan)
 * ASTERISK-24241 - crash: CDRs recursively attempt to update Party
      B information in a multi-party bridge, overrunning the stack
      (Reported by Deepak Singh Rawat)
 * ASTERISK-24254 - CDRs: Application/args/dialplan CEP updated
      during dial operation (Reported by Matt Jordan)
 * ASTERISK-24231 - crash: CLI execution of realtime destroy
      sippeers id 1 causes crash due to NULL name provided to
      ast_variable (Reported by Niklas Larsson)
 * ASTERISK-24249 - SIP debugs do not stop (Reported by Avinash
      Mohod)
 * ASTERISK-23577 - res_rtp_asterisk: Crash in
      ast_rtp_on_turn_rtp_state when RTP instance is NULL (Reported by
      Jay Jideliov)
 * ASTERISK-23634 - With TURN Asterisk crashes on multiple (7-10)
      concurrent WebRTC (avpg/encryption/icesupport) calls (Reported
      by Roman Skvirsky)
 * ASTERISK-24161 - PJSIPShowEndpoint gives inaccurate count of
      list items (Reported by Mark Michelson)
 * ASTERISK-24331 - Unexpected Errors in Asterisk Manager Interface
      Output (Reported by xrobau)
 * ASTERISK-24136 - Security: Crash in Asterisk's PJSIP code when
      subscribing to an event with an unexpected body type (Reported
      by Mark Michelson)
 * ASTERISK-24301 - Security: Out of call MESSAGE requests
      processed via Message channel driver can crash Asterisk
      (Reported by Matt Jordan)
 * ASTERISK-24290 - Endpoint identifier match value fails to parse
      when CIDR network format is specified (Reported by Ray Crumrine)
 * ASTERISK-24237 - CDR: FRACK With PJSIP blonde transfer.
      (Reported by Richard Mudgett)

Improvements made in this release:
-----------------------------------
 * ASTERISK-24171 - [patch] Provide a manpage for the aelparse
      utility (Reported by Jeremy Lain??)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.6.0

Thank you for your continued support of Asterisk!