<div dir="ltr">Nova versão do Asterisk 13.6.0<div><br></div><div>- Muitos bugs foram corrigidos.</div><div><div class="gmail_quote"><br></div><div class="gmail_quote"><br>The Asterisk Development Team has announced the release of Asterisk 13.6.0.<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 13.6.0 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-25377 - res_pjsip: Change default "From user" from UUID<br>
to something more palatable (Reported by Mark Michelson)<br>
* ASTERISK-25252 - ARI: Add the ability to manipulate log channels<br>
(Reported by Matt Jordan)<br>
<br>
Bugs fixed in this release:<br>
-----------------------------------<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-25438 - res_rtp_asterisk: ICE role message even when<br>
ICE is not enabled (Reported by Joshua Colp)<br>
* ASTERISK-25383 - Core dumps on startup and shutdown with<br>
MALLOC_DEBUG enabled (Reported by yaron nahum)<br>
* ASTERISK-25423 - Caller gets no Connected line update during<br>
call pickup. (Reported by Richard Mudgett)<br>
* ASTERISK-25305 - Dynamic logger channels can be added multiple<br>
times (Reported by Mark Michelson)<br>
* ASTERISK-25418 - On-hold channels redirected out of a bridge<br>
appear to still be on hold (Reported by Mark Michelson)<br>
* ASTERISK-25384 - Regular Asterisk crashes when using Page<br>
application. "user_data is NULL" (Reported by Chet Stevens)<br>
* ASTERISK-25407 - Asterisk fails to log to multiple syslog<br>
destinations (Reported by Elazar Broad)<br>
* ASTERISK-25410 - app_record: RECORDED_FILE variable not being<br>
populated (Reported by Kevin Harwell)<br>
* ASTERISK-25394 - pbx: Incorrect device and presence state when<br>
changing hint details (Reported by Joshua Colp)<br>
* ASTERISK-25396 - chan_sip: Extremely long callerid name causes<br>
invalid SIP (Reported by Walter Doekes)<br>
* ASTERISK-25399 - app_queue: AgentComplete event has wrong reason<br>
(Reported by Kevin Harwell)<br>
* ASTERISK-25185 - Segfault in app_queue on transfer scenarios<br>
(Reported by Etienne Lessard)<br>
* ASTERISK-25353 - [patch] Transcoding while different in Frame<br>
size = Frames lost (Reported by Alexander Traud)<br>
* ASTERISK-25325 - ARI PUT reload chan_sip HTTP response 404<br>
(Reported by Rodrigo Ramirez Norambuena)<br>
* ASTERISK-25390 - default_from_user can crash with certain<br>
configuration backends (Reported by Mark Michelson)<br>
* ASTERISK-25387 - res_pjsip_nat: Malformed REGISTER request<br>
causes NAT'd Contact header to not be rewritten (Reported by<br>
Matt Jordan)<br>
* ASTERISK-25227 - No audio at in-band announcements in ooh323<br>
channel (Reported by Alexandr Dranchuk)<br>
* ASTERISK-25369 - res_parking: ParkAndAnnounce - Inheritable<br>
variables aren't applied to the announcer channel (Reported by<br>
Jonathan Rose)<br>
* ASTERISK-25295 - res_pjsip crash - pjsip_uri_get_uri at<br>
/usr/include/pjsip/sip_uri.h (Reported by Dmitriy Serov)<br>
* ASTERISK-25381 - res_pjsip: AoRs deleted via ARI (or other<br>
mechanism) do not destroy their related contacts (Reported by<br>
Matt Jordan)<br>
* ASTERISK-25367 - pbx: Long pattern match hints may cause "core<br>
show hints" to crash (Reported by Joshua Colp)<br>
* ASTERISK-25365 - Persistent subscriptions have extra<br>
Content-Length/corrupted messages (Reported by Mark Michelson)<br>
* ASTERISK-25362 - Deadlock due to presence state callback<br>
(Reported by Mark Michelson)<br>
* ASTERISK-25356 - res_pjsip_sdp_rtp: Multiple keepalive scheduled<br>
items may exist (Reported by Joshua Colp)<br>
* ASTERISK-25355 - sched: ast_sched_del may return prematurely due<br>
to spurious wakeup (Reported by Joshua Colp)<br>
* ASTERISK-25318 -<br>
tests/rest_api/applications/subscribe-endpoint/nominal/resource:<br>
Sporadically failing (Reported by Joshua Colp)<br>
* ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup<br>
cause on call pickup (Reported by Joshua Colp)<br>
* ASTERISK-25342 - res_pjsip: Repeated usage of pj_gethostip may<br>
block (Reported by Joshua Colp)<br>
* ASTERISK-25341 - bridge: Hangups may get lost when executing<br>
actions (Reported by Joshua Colp)<br>
* ASTERISK-25339 - res_pjsip: Empty "auth" sections from<br>
non-config backgrounds are interpreted as valid (Reported by<br>
Matt Jordan)<br>
* ASTERISK-25215 - Differences in queue.log between Set<br>
QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne<br>
Gaetz)<br>
* ASTERISK-25322 - Crash occurs when using MixMonitor with t() or<br>
r() options. (Reported by Richard Mudgett)<br>
* ASTERISK-25320 - chan_sip.c: sip_report_security_event searches<br>
for wrong or non existent peer on invite (Reported by Kevin<br>
Harwell)<br>
* ASTERISK-25315 - DAHDI channels send shortened duration DTMF<br>
tones. (Reported by Richard Mudgett)<br>
* ASTERISK-25312 - res_http_websocket: Terminate connection on<br>
fatal cases (Reported by Joshua Colp)<br>
* ASTERISK-25306 - Persistent subscriptions can save multiple SIP<br>
messages at once, leading to potential crashes. (Reported by<br>
Mark Michelson)<br>
* ASTERISK-25309 - [patch] iLBC 20 advertised (Reported by<br>
Alexander Traud)<br>
* ASTERISK-25304 - res_pjsip: XML sanitization may write past<br>
buffer (Reported by Joshua Colp)<br>
* ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on<br>
Firefox 39 - add ECDH support and fallback to prime256v1<br>
(Reported by Stefan Engström)<br>
* ASTERISK-25296 - RTP performance issue with several channel<br>
drivers. (Reported by Richard Mudgett)<br>
* ASTERISK-25297 - Crashes running<br>
channels/pjsip/resolver/srv/failover/in_dialog testsuite tests<br>
(Reported by Richard Mudgett)<br>
* ASTERISK-25292 - Testuite:<br>
tests/apps/bridge/bridge_wait/bridge_wait_e_options fails<br>
(Reported by Kevin Harwell)<br>
* ASTERISK-25271 - Parking & blind transfer: Transferer channel<br>
not hung up if no MOH (Reported by Kevin Harwell)<br>
<br>
Improvements made in this release:<br>
-----------------------------------<br>
* ASTERISK-24870 - ARI: Subscriptions to bridges generally not<br>
super useful (Reported by Matt Jordan)<br>
* ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()<br>
defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)<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-13.6.0" rel="noreferrer" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.6.0</a><br>
<br>
Thank you for your continued support of Asterisk!<br>
<br>
<br>--<br>
_____________________________________________________________________<br>
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