[AsteriskBrasil] Fwd: [asterisk-dev] Asterisk 12.7.0 Now Available

Segunda Novembro 10 16:09:42 BRST 2014

The Asterisk Development Team has announced the release of Asterisk 12.7.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 12.7.0 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-24339 - Swagger API Docs have incorrect basePath
      (Reported by Bradley Watkins)
 * 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-24295 - crash: creating out of dialog OPTIONS request
      crashes (Reported by Rogger Padilla)
 * 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-24350 - PJSIP shows commands prints unneeded headers
      (Reported by snuffy)
 * ASTERISK-22945 - [patch] Memory leaks in chan_sip.c with
      realtime peers (Reported by ibercom)
 * ASTERISK-24362 - res_hep leaks reference to configuration
      (Reported by Corey Farrell)
 * ASTERISK-23781 - outgoing missing as enum from
      contrib/ast-db-manage/config (Reported by Stephen More)
 * ASTERISK-24199 - 'ALL' is specified in pjsip.conf.sample for TLS
      cipher but it is not valid (Reported by Joshua Colp)
 * ASTERISK-24262 - AMI CoreShowChannel missing several output
      fields and event documentation (Reported by Mitch Claborn)
 * ASTERISK-24356 - PJSIP: Directed pickup causes deadlock
      (Reported by Richard Mudgett)
 * ASTERISK-24195 - bridge_native_rtp: Removing mixmonitor from a
      native RTP capable smart bridge doesn't cause the bridge to
      resume being a native rtp bridge (Reported by Jonathan Rose)
 * 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-24369 - res_pjsip: Large message on reliable transport
      can cause empty messages to be passed from the PJSIP stack up,
      causing crashes in multiple locations (Reported by Matt Jordan)
 * ASTERISK-24382 - chan_pjsip: Calling PJSIP_MEDIA_OFFER on a
      non-PJSIP channel results in an invalid reference of a channel
      pvt and a FRACK (Reported by Matt Jordan)
 * ASTERISK-24370 - res_pjsip/pjsip_options: OPTIONS request sent
      to Asterisk with no user in request is always 404'd (Reported by
      Matt Jordan)
 * ASTERISK-24224 - When using Bridge() dialplan application,
      surrogate channel appears in list and call count is inflated.
      (Reported by Mark Michelson)
 * ASTERISK-24354 - AMI sendMessage closes AMI connection on error
      (Reported by Peter Katzmann)
 * ASTERISK-24398 - Initialize auth_rejection_permanent on client
      state to the configuration parameter value (Reported by Matt
      Jordan)
 * 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-24387 - res_pjsip: rport sent from UAS MUST include the
      port that the UAC sent the request on (Reported by Matt Jordan)
 * 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-24394 - CDR: FRACK with PJSIP directed pickup.
      (Reported by Richard Mudgett)
 * ASTERISK-24392 - res_fax: fax gateway sessions leak (Reported by
      Corey Farrell)
 * ASTERISK-24321 - SIP deadlock when running automated queues
      tests (Reported by Steve Pitts)
 * 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-24312 - SIGABRT when improperly configured realtime
      pjsip  (Reported by Dafi Ni)
 * ASTERISK-24426 - CDR Batch mode: size used as time value after
      first expire (Reported by Shane Blaser)
 * ASTERISK-24327 - bridge_native_rtp: Smart bridge operation to
      softmix sometimes fails to properly re-INVITE remotely bridged
      participants (Reported by Matt Jordan)
 * ASTERISK-24415 - Missing AMI VarSet events when channels inherit
      variables. (Reported by Richard Mudgett)
 * ASTERISK-24063 - [patch]Asterisk does not respect outbound proxy
      when sending qualify requests (Reported by Damian Ivereigh)
 * ASTERISK-24122 - Documentaton for res_pjsip option use_avpf
      needs to be fixed (Reported by James Van Vleet)
 * ASTERISK-24381 - res_pjsip_sdp_rtp: Declined media streams are
      interpreted, leading to erroneous 488 rejections (Reported by
      Matt Jordan)
 * 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-24437 - Review implementation of ast_bridge_impart for
      leaks and document proper usage (Reported by Scott Griepentrog)
 * ASTERISK-24453 - manager: acl_change_sub leaks (Reported by
      Corey Farrell)
 * 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-24462 - res_pjsip: Stale qualify statistics after
      disablementation (Reported by Kevin Harwell)
 * 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-24411 - [patch] Status of outbound registration is not
      changed upon unregistering. (Reported by John Bigelow)
 * ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream
      leaks (Reported by Corey Farrell)
 * ASTERISK-24487 - configuration: sections should be loadable as
      template even when not marked (Reported by Scott Griepentrog)
 * ASTERISK-24307 - Unintentional memory retention in stringfields
      (Reported by Etienne Lessard)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.7.0

Thank you for your continued support of Asterisk!

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Sylvio Jollenbeck
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