<div class="gmail_quote">---------- Mensagem encaminhada ----------<br>De: "Asterisk Development Team" <<a href="mailto:asteriskteam@digium.com">asteriskteam@digium.com</a>><br>Data: 15/01/2016 7:23 PM<br>Assunto: [asterisk-dev] Asterisk 11.21.0 Now Available<br>Para: "Asterisk Developers Mailing List" <<a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br>Cc: <br><br type="attribution">The Asterisk Development Team has announced the release of Asterisk 11.21.0.<br>
This release is available for immediate download at<br>
<a href="http://downloads.asterisk.org/pub/telephony/asterisk" rel="noreferrer" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk</a><br>
<br>
The release of Asterisk 11.21.0 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-25640 - pbx: Deadlock on features reload and state<br>
change hint. (Reported by Krzysztof Trempala)<br>
* ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and<br>
thread of asterisk is not released (Reported by Hiroaki Komatsu)<br>
* ASTERISK-25569 - app_meetme: Audio quality issues (Reported by<br>
Corey Farrell)<br>
* ASTERISK-25609 - [patch]Asterisk may crash when calling<br>
ast_channel_get_t38_state(c) (Reported by Filip Jenicek)<br>
* ASTERISK-24146 - [patch]No audio on WebRtc caller side when<br>
answer waiting time is more than ~7sec (Reported by Aleksei<br>
Kulakov)<br>
* ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec<br>
(Reported by Alexander Traud)<br>
* ASTERISK-25616 - Warning with a Codec Module which supports PLC<br>
with FEC (Reported by Alexander Traud)<br>
* ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by<br>
Dudás József)<br>
* ASTERISK-25498 - Asterisk crashes when negotiating g729 without<br>
that module installed (Reported by Ben Langfeld)<br>
* ASTERISK-25476 - chan_sip loses registrations after a while<br>
(Reported by Michael Keuter)<br>
* ASTERISK-25593 - fastagi: record file closed after sending<br>
result (Reported by Kevin Harwell)<br>
* ASTERISK-25585 - [patch]rasterisk never hits most of main(), but<br>
it's assumed to (Reported by Walter Doekes)<br>
* ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by<br>
Joshua Colp)<br>
* ASTERISK-25449 - main/sched: Regression introduced by<br>
5c713fdf18f causes erroneous duplicate RTCP messages; other<br>
potential scheduling issues in chan_sip/chan_skinny (Reported by<br>
Matt Jordan)<br>
* ASTERISK-25537 - [patch] format-attribute module: RFC or<br>
internal defaults? (Reported by Alexander Traud)<br>
* ASTERISK-25373 - add documentation for CALLERID(pres) and also<br>
the CONNECTEDLINE and REDIRECTING variants (Reported by Walter<br>
Doekes)<br>
* ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by<br>
Walter Doekes)<br>
* ASTERISK-25434 - Compiler flags not reported in 'core show<br>
settings' despite usage during compilation (Reported by Rusty<br>
Newton)<br>
* ASTERISK-25494 - build: GCC 5.1.x catches some new const, array<br>
bounds and missing paren issues (Reported by George Joseph)<br>
* ASTERISK-7803 - [patch] Update the maximum packetization values<br>
in frame.c (Reported by dea)<br>
* ASTERISK-25461 - Nested dialplan #includes don't work as<br>
expected. (Reported by Richard Mudgett)<br>
* ASTERISK-25455 - Deadlock of PJSIP realtime over<br>
res_config_pgsql (Reported by mdu113)<br>
* ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing<br>
(Reported by Olle Johansson)<br>
* ASTERISK-25400 - Hints broken when "CustomPresence" doesn't<br>
exist in AstDB (Reported by Andrew Nagy)<br>
* ASTERISK-25443 - [patch]IPv6 - Potential issue in via header<br>
parsing (Reported by ffs)<br>
* ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON<br>
(Reported by Bojan Nemčić)<br>
* ASTERISK-25438 - res_rtp_asterisk: ICE role message even when<br>
ICE is not enabled (Reported by Joshua Colp)<br>
<br>
Improvements made in this release:<br>
-----------------------------------<br>
* ASTERISK-24718 - [patch]Add inital support of "sanitize" to<br>
configure (Reported by Badalian Vyacheslav)<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-11.21.0" rel="noreferrer" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0</a><br>
<br>
Thank you for your continued support of Asterisk!<br>
<br>
<br>--<br>
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