[asterisk-dev] Asterisk 12.8.0-rc1 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the first release candidate of
Asterisk 12.8.0. This release candidate is available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 12.8.0-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-24480 - res_http_websockets: Module reference decrease
      below zero (Reported by Corey Farrell)
 * ASTERISK-24482 - func_talkdetect: Fix stasis message leak in
      audiohook callback (Reported by Corey Farrell)
 * ASTERISK-24487 - configuration: sections should be loadable as
      template even when not marked (Reported by Scott Griepentrog)
 * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
      unescapes semicolons ("\;" -> ";") turning them into comments
      (corruption) on rewrite of a config file (Reported by George
      Joseph)
 * ASTERISK-24438 - res_pjsip_multihomed.so blocks Asterisk reload
      when DNS settings invalid (Reported by Melissa Shepherd)
 * ASTERISK-24307 - Unintentional memory retention in stringfields
      (Reported by Etienne Lessard)
 * ASTERISK-24491 - Memory leak in res_hep (Reported by Zane
      Conkle)
 * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
      extra calls to ast_module_unref (Reported by Corey Farrell)
 * ASTERISK-24447 - Bridge DTMF hooks: Audio doesn't pass when
      waiting for more matching digits. (Reported by Richard Mudgett)
 * ASTERISK-24257 - agent must dial acceptdtmf twice to bridge to
      queue caller (Reported by Steve Pitts)
 * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
      (Reported by Corey Farrell)
 * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
      length exceeds 50 (roughly) national symbols (Reported by
      Dmitriy Bubnov)
 * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
      revision r227276 (Reported by Xavier Hienne)
 * ASTERISK-24505 - manager: http connections leak references
      (Reported by Corey Farrell)
 * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
      coverage are enabled (Reported by Corey Farrell)
 * ASTERISK-24444 - PBX: Crash when generating extension for
      pattern matching hint (Reported by Leandro Dardini)
 * ASTERISK-24489 - Crash: Asterisk crashes when converting RTCP
      packet to JSON for res_hep_rtcp and report blocks are greater
      than 1 (Reported by Gregory Malsack)
 * ASTERISK-24498 - Segmentation fault in res_hep_rtcp on attended
      transfer (Reported by Beppo Mazzucato)
 * ASTERISK-24501 - ARI: Moving a channel between bridges followed
      by a hangup can cause an ARI client to not receive an expected
      ChannelLeftBridge event before StasisEnd (Reported by Matt
      Jordan)
 * ASTERISK-24336 - PJSIP timer_min_se value under 90 causes crash
      (Reported by Leon Rowland)
 * ASTERISK-23651 - Reloading some modules that are loaded already,
      results in 'No such module' before a successful reload (Reported
      by Rusty Newton)
 * ASTERISK-24499 - Need more explicit debug when PJSIP dialstring
      is invalid (Reported by Rusty Newton)
 * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
      endmarked users when last marked user leaves (Reported by Matt
      Jordan)
 * ASTERISK-15242 - transmit_refer leaks sip_refer structures
      (Reported by David Woolley)
 * ASTERISK-24508 - pjsip - REFER request from SNOM is rejected
      with "400 bad request" - DEBUG shows "Received a REFER without a
      parseable Refer-To" (Reported by Beppo Mazzucato)
 * ASTERISK-24535 - stringfields: Fix regression from fix for
      unintentional memory retention and another issue exposed by the
      fix (Reported by Corey Farrell)
 * ASTERISK-24471 - Crash - assert_fail in libc in
      pjmedia_sdp_neg_negotiate from /usr/local/lib/libpjmedia.so.2
      (Reported by yaron nahum)
 * ASTERISK-24528 - res_pjsip_refer: Sending INVITE with Replaces
      in-dialog with invalid target causes crash (Reported by Joshua
      Colp)
 * ASTERISK-24531 - res_pjsip_acl: ACLs not applied on initial
      module load (Reported by Matt Jordan)
 * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
      allow blocked addresses through (Reported by Matt Jordan)
 * ASTERISK-24533 - 2 threads created per chan_sip entry (Reported
      by xrobau)
 * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
      voicemail under high concurrency with an IMAP backend (Reported
      by David Duncan Ross Palmer)
 * ASTERISK-24572 - [patch]App_meetme is loaded without its
      defaults when the configuration file is missing (Reported by
      Nuno Borges)
 * ASTERISK-24573 - [patch]Out of sync conversation recording when
      divided in multiple recordings (Reported by Nuno Borges)
 * ASTERISK-24537 - Stasis: StasisStart/StasisEnd events are not
      reliably transmitted during transfers (Reported by Matt Jordan)

Improvements made in this release:
-----------------------------------
 * ASTERISK-24279 - Documentation: Clarify the behaviour of the CDR
      property 'unanswered' (Reported by Matt Jordan)
 * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
      column in the cel_odbc module (Reported by Etienne Lessard)
 * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
      recordings (Reported by Ben Smithurst)
 * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
      lookups (Reported by Birger "WIMPy" Harzenetter)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.8.0-rc1

Thank you for your continued support of Asterisk!