<div dir="ltr"><br><br><div class="gmail_quote">---------- Forwarded message ----------<br>From: <b class="gmail_sendername">Asterisk Development Team</b> <span dir="ltr"><<a href="mailto:asteriskteam@digium.com">asteriskteam@digium.com</a>></span><br>
Date: 2014-03-03 18:42 GMT-03:00<br>Subject: [asterisk-dev] Asterisk 11.8.0 Now Available<br>To: Asterisk Developers Mailing List <<a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br>
<br><br>The Asterisk Development Team has announced the release of Asterisk 11.8.0.<br>
This release is available for immediate download at<br>
<a href="http://downloads.asterisk.org/pub/telephony/asterisk" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk</a><br>
<br>
The release of Asterisk 11.8.0 resolves several issues reported by the<br>
community and would have not been possible without your participation.<br>
Thank you!<br>
<br>
The following are the issues resolved in this release:<br>
<br>
Bugs fixed in this release:<br>
-----------------------------------<br>
* ASTERISK-22544 - Italian prompt vm-options has advertisement in<br>
it (Reported by Rusty Newton)<br>
* ASTERISK-21383 - STUN Binding Requests Not Being Sent Back from<br>
Asterisk to Chrome (Reported by Shaun Clark)<br>
* ASTERISK-22478 - [patch]Can't use pound(hash) symbol for custom<br>
DTMF menus in ConfBridge (processed as directive) (Reported by<br>
Nicolas Tanski)<br>
* ASTERISK-12117 - chan_sip creates a new local tag (from-tag) for<br>
every register message (Reported by Pawel Pierscionek)<br>
* ASTERISK-20862 - Asterisk min and max member penalties not<br>
honored when set with 0 (Reported by Schmooze Com)<br>
* ASTERISK-22746 - [patch]Crash in chan_dahdi during caller id<br>
read (Reported by Michael Walton)<br>
* ASTERISK-22788 - [patch] main/translate.c: access to variable f<br>
after free in ast_translate() (Reported by Corey Farrell)<br>
* ASTERISK-21242 - Segfault when T.38 re-invite retransmission<br>
receives 200 OK (Reported by Ashley Winters)<br>
* ASTERISK-22590 - BufferOverflow in unpacksms16() when receiving<br>
16 bit multipart SMS with app_sms (Reported by Jan Juergens)<br>
* ASTERISK-22905 - Prevent Asterisk functions that are 'dangerous'<br>
from being executed from external interfaces (Reported by Matt<br>
Jordan)<br>
* ASTERISK-23021 - Typos in code : "avaliable" instead of<br>
"available" (Reported by Jeremy Lainé)<br>
* ASTERISK-22970 - [patch]Documentation fix for QUOTE() (Reported<br>
by Gareth Palmer)<br>
* ASTERISK-21960 - ooh323 channels stuck (Reported by Dmitry<br>
Melekhov)<br>
* ASTERISK-22350 - DUNDI - core dump on shutdown - segfault in<br>
sqlite3_reset from /usr/lib/libsqlite3.so.0 (Reported by Birger<br>
"WIMPy" Harzenetter)<br>
* ASTERISK-22942 - [patch] - Asterisk crashed after<br>
Set(FAXOPT(faxdetect)=t38) (Reported by adomjan)<br>
* ASTERISK-22856 - [patch]SayUnixTime in polish reads minutes<br>
instead of seconds (Reported by Robert Mordec)<br>
* ASTERISK-22854 - [patch] - Deadlock between cel_pgsql unload and<br>
core_event_dispatcher taskprocessor thread (Reported by Etienne<br>
Lessard)<br>
* ASTERISK-22910 - [patch] - REPLACE() calls strcpy on overlapping<br>
memory when <replace-char> is empty (Reported by Gareth Palmer)<br>
* ASTERISK-22871 - cel_pgsql module not loading after "reload" or<br>
"reload cel_pgsql.so" command (Reported by Matteo)<br>
* ASTERISK-23084 - [patch]rasterisk needlessly prints the<br>
AST-2013-007 warning (Reported by Tzafrir Cohen)<br>
* ASTERISK-17138 - [patch] Asterisk not re-registering after it<br>
receives "Forbidden - wrong password on authentication"<br>
(Reported by Rudi)<br>
* ASTERISK-23011 - [patch]<a href="http://configure.ac" target="_blank">configure.ac</a> and pbx_lua don't support<br>
lua 5.2 (Reported by George Joseph)<br>
* ASTERISK-22834 - Parking by blind transfer when lot full orphans<br>
channels (Reported by rsw686)<br>
* ASTERISK-23047 - Orphaned (stuck) channel occurs during a failed<br>
SIP transfer to parking space (Reported by Tommy Thompson)<br>
* ASTERISK-22946 - Local From tag regression with <a href="http://sipgate.de" target="_blank">sipgate.de</a><br>
(Reported by Stephan Eisvogel)<br>
* ASTERISK-23010 - No BYE message sent when sip INVITE is received<br>
(Reported by Ryan Tilton)<br>
* ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set<br>
- probably introduced in 11.7.0 (Reported by OK)<br>
<br>
Improvements made in this release:<br>
-----------------------------------<br>
* ASTERISK-22728 - [patch] Improve Understanding Of 'Forcerport'<br>
When Running "sip show peers" (Reported by Michael L. Young)<br>
* ASTERISK-22659 - Make a new core and extra sounds release<br>
(Reported by Rusty Newton)<br>
* ASTERISK-22919 - core show channeltypes slicing (Reported by<br>
outtolunc)<br>
* ASTERISK-22918 - dahdi show channels slices PRI channel dnid on<br>
output (Reported by outtolunc)<br>
<br>
For a full list of changes in this release, please see the ChangeLog:<br>
<br>
<a href="http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.8.0" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.8.0</a><br>
<br>
Thank you for your continued support of Asterisk!<br>
<br>
<br>--<br>
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