<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 7:27 PM<br>Assunto: [asterisk-dev] Asterisk 13.7.0 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 13.7.0.<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 13.7.0 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>
New Features made in this release:<br>
-----------------------------------<br>
* ASTERISK-25419 - Dialplan Application for Integration of StatsD<br>
(Reported by Ashley Sanders)<br>
* ASTERISK-25549 - Confbridge: Add participant timeout option<br>
(Reported by Mark Michelson)<br>
* ASTERISK-24922 - ARI: Add the ability to intercept hold and<br>
raise an event (Reported by Matt Jordan)<br>
<br>
Bugs fixed in this release:<br>
-----------------------------------<br>
* ASTERISK-25689 - pjsip show contacts not working in Asterisk<br>
13.7rc2 (Reported by Marcelo Terres)<br>
* ASTERISK-25640 - pbx: Deadlock on features reload and state<br>
change hint. (Reported by Krzysztof Trempala)<br>
* ASTERISK-25664 - ast_format_cap_append_by_type leaks a reference<br>
(Reported by Corey Farrell)<br>
* ASTERISK-25601 - json: Audit reference usage and thread safety<br>
(Reported by Joshua Colp)<br>
* ASTERISK-25625 - res_sorcery_memory_cache: Add full backend<br>
caching (Reported by Joshua Colp)<br>
* ASTERISK-25615 - res_pjsip: Setting transport async_operations ><br>
1 causes segfault on tls transports (Reported by George Joseph)<br>
* ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and<br>
thread of asterisk is not released (Reported by Hiroaki Komatsu)<br>
* ASTERISK-25619 - res_chan_stats not sending the correct<br>
information to StatsD (Reported by Tyler Cambron)<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-25608 - res_pjsip/contacts/statsd: Lifecycle events<br>
aren't consistent (Reported by George Joseph)<br>
* ASTERISK-25584 - [patch] format-attribute module: VP8 missing<br>
(Reported by Alexander Traud)<br>
* ASTERISK-25583 - [patch] format-attribute module: RFC 7587 (Opus<br>
Codec) (Reported by Alexander Traud)<br>
* ASTERISK-25498 - Asterisk crashes when negotiating g729 without<br>
that module installed (Reported by Ben Langfeld)<br>
* ASTERISK-25595 - Unescaped : in messge sent to statsd (Reported<br>
by Niklas Larsson)<br>
* ASTERISK-25476 - chan_sip loses registrations after a while<br>
(Reported by Michael Keuter)<br>
* ASTERISK-25598 - res_pjsip: Contact status messages are<br>
printing a hash instead of the uri (Reported by George Joseph)<br>
* ASTERISK-25600 - bridging: Inconsistency in BRIDGEPEER (Reported<br>
by Jonathan Rose)<br>
* ASTERISK-25582 - Testsuite: Reactor timeout error in<br>
tests/fax/pjsip/directmedia_reinvite_t38 (Reported by Matt<br>
Jordan)<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-25590 - CLI Usage info for 'pjsip send notify'<br>
references incorrect config (Reported by Corey Farrell)<br>
* ASTERISK-25165 - Testsuite - Sorcery memory cache leaks<br>
(Reported by Corey Farrell)<br>
* ASTERISK-25575 - res_pjsip: Dynamic outbound registrations<br>
created via ARI are not loaded into memory on Asterisk<br>
start/restart (Reported by Matt Jordan)<br>
* ASTERISK-25545 - [patch] translation module gets cached not<br>
joint format (Reported by Alexander Traud)<br>
* ASTERISK-25573 - [patch] H.264 format attribute module: resets<br>
whole SDP (Reported by Alexander Traud)<br>
* ASTERISK-24958 - Forwarding loop detection inhibits certain<br>
desirable scenarios (Reported by Mark Michelson)<br>
* ASTERISK-25561 - app_queue.c line 6503 (try_calling): mutex<br>
'qe->chan' freed more times than we've locked! (Reported by Alec<br>
Davis)<br>
* ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by<br>
Joshua Colp)<br>
* ASTERISK-25160 - [patch] Opus Codec: SIP/SDP line fmtp missing<br>
when called internally (Reported by Alexander Traud)<br>
* ASTERISK-25535 - [patch] format creation on module load instead<br>
of cache (Reported by Alexander Traud)<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-25546 - threadpool: Race condition between idle timeout<br>
and activation (Reported by Joshua Colp)<br>
* ASTERISK-25537 - [patch] format-attribute module: RFC or<br>
internal defaults? (Reported by Alexander Traud)<br>
* ASTERISK-25533 - [patch] buffer for ast_format_cap_get_names<br>
only 64 bytes (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-24779 - Passthrough OPUS codec not working with<br>
chan_pjsip (Reported by PowerPBX)<br>
* ASTERISK-25522 - ARI: Crash when creating channel via ARI<br>
originate with requesting channel (Reported by Matt Jordan)<br>
* ASTERISK-25434 - Compiler flags not reported in 'core show<br>
settings' despite usage during compilation (Reported by Rusty<br>
Newton)<br>
* ASTERISK-24106 - WebSockets Automatically decides what driver it<br>
will use (Reported by Andrew Nagy)<br>
* ASTERISK-25513 - Crash: malloc failed with high load of<br>
subscriptions. (Reported by John Bigelow)<br>
* ASTERISK-25505 - res_pjsip_pubsub: Crash on off-nominal when UAS<br>
dialog can't be created (Reported by Joshua Colp)<br>
* ASTERISK-24543 - Asterisk 13 responds to SIP Invite with all<br>
possible codecs configured for peer as opposed to intersection<br>
of configured codecs and offered codecs (Reported by Taylor<br>
Hawkes)<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-25485 - res_pjsip_outbound_registration: registration<br>
stops due to 400 response (Reported by Kevin Harwell)<br>
* ASTERISK-25486 - res_pjsip: Fix deadlock when validating URIs<br>
(Reported by Joshua Colp)<br>
* ASTERISK-7803 - [patch] Update the maximum packetization values<br>
in frame.c (Reported by dea)<br>
* ASTERISK-25484 - [patch] autoframing=yes has no effect (Reported<br>
by Alexander Traud)<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-25435 - Asterisk periodically hangs. UDP Recv-Q greatly<br>
exceeds zero. (Reported by Dmitriy Serov)<br>
* ASTERISK-25451 - Broken video - erased rtp marker bit (Reported<br>
by Stefan Engström)<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-25404 - segfault/crash in chan_pjsip_hangup ... at<br>
chan_pjsip.c (Reported by Chet Stevens)<br>
* ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON<br>
(Reported by Bojan Nemčić)<br>
* ASTERISK-25441 - Deadlock in res_sorcery_memory_cache. (Reported<br>
by Richard Mudgett)<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-25618 - res_pjsip: Check for readability of TLS files<br>
at startup (Reported by George Joseph)<br>
* ASTERISK-25572 - Endpoints: Add StatsD stats for Asterisk<br>
endpoints (Reported by Matt Jordan)<br>
* ASTERISK-25571 - PJSIP: Add StatsD stats for some common PJSIP<br>
objects (Reported by Matt Jordan)<br>
* ASTERISK-25518 - taskprocessor: Add high water mark (Reported by<br>
Jonathan Rose)<br>
* ASTERISK-25477 - pjsip show "command" like [criteria] (Reported<br>
by Bryant Zimmerman)<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-13.7.0" rel="noreferrer" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.7.0</a><br>
<br>
Thank you for your continued support of Asterisk!<br>
<br>
<br>--<br>
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