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[asteriskteam at digium...]

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

The release of Asterisk 12.8.0 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-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-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, please see the ChangeLog:

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

Thank you for your continued support of Asterisk!

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