<div dir="ltr">PSC<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: 2014-05-29 17:04 GMT-03:00<br>Subject: [asterisk-dev] Asterisk 12.3.0 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 12.3.0.<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 12.3.0 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>
Improvements made in this release:<br>
-----------------------------------<br>
* ASTERISK-23553 - Add ast_spinlock capability to lock.h (Reported<br>
by George Joseph)<br>
* ASTERISK-23649 - [patch]Support for DTLS retransmission<br>
(Reported by NITESH BANSAL)<br>
* ASTERISK-23564 - [patch]TLS/SRTP status of channel not currently<br>
available in a CLI command (Reported by Patrick Laimbock)<br>
* ASTERISK-23754 - [patch] Use var/lib directory for log file<br>
configured in asterisk.conf (Reported by Igor Goncharovsky)<br>
<br>
Bugs fixed in this release:<br>
-----------------------------------<br>
* ASTERISK-23547 - [patch] app_queue removing callers from queue<br>
when reloading (Reported by Italo Rossi)<br>
* ASTERISK-22846 - testsuite: masquerade super test fails on all<br>
branches (still) (Reported by Matt Jordan)<br>
* ASTERISK-23390 - NewExten Event with application AGI shows up<br>
before and after AGI runs (Reported by Benjamin Keith Ford)<br>
* ASTERISK-23584 - PJSIP 'Unable to create channel' when<br>
attempting to call from endpoint with UDP transport to one using<br>
WebSockets (Reported by Rusty Newton)<br>
* ASTERISK-23545 - Confbridge talker detection settings<br>
configuration load bug (Reported by John Knott)<br>
* ASTERISK-23546 - CB_ADD_LEN does not do what you'd think<br>
(Reported by Walter Doekes)<br>
* ASTERISK-22904 - bridges: lock the bridge when creating bridge<br>
snapshots (Reported by Matt Jordan)<br>
* ASTERISK-23620 - Code path in app_stack fails to unlock list<br>
(Reported by Bradley Watkins)<br>
* ASTERISK-23616 - Big memory leak in logger.c (Reported by<br>
ibercom)<br>
* ASTERISK-23588 - ARI: Crash when unsubscribing from bridge<br>
(Reported by Matt Jordan)<br>
* ASTERISK-23502 - Channel variable SIPREFERTOHDR not being set<br>
during blind transfer (Reported by John Bigelow)<br>
* ASTERISK-23576 - Build failure on SmartOS / Illumos / SunOS<br>
(Reported by Sebastian Wiedenroth)<br>
* ASTERISK-23514 - The pjsip.conf aor qualify contact parameters<br>
are not updated on reload. (Reported by Richard Mudgett)<br>
* ASTERISK-23550 - Newer sound sets don't show up in menuselect<br>
(Reported by Rusty Newton)<br>
* ASTERISK-22677 - Playbacks on bridge via ARI are not queued<br>
(Reported by John Bigelow)<br>
* ASTERISK-18331 - app_sms failure (Reported by David Woodhouse)<br>
* ASTERISK-23487 - features.conf cant load from realtime because<br>
features_config.c starts before loader.c (Reported by Denis)<br>
* ASTERISK-23282 - Documentation - Tab completion and CLI usage<br>
documentation do not indicate that 'all' is accepted for<br>
'confbridge kick all' (Reported by Dorian Logan)<br>
* ASTERISK-19465 - P-Asserted-Identity Privacy (Reported by<br>
Krzysztof Chmielewski)<br>
* ASTERISK-23573 - Crash when transferring unbridged call - in<br>
bridge_app_subscribed at stasis/app.c (Reported by Mark<br>
Michelson)<br>
* ASTERISK-23639 - PJSIP Realtime: Alembic migration needed in<br>
order to widen some string columns (Reported by Mark Michelson)<br>
* ASTERISK-23560 - [ARI] MOH doesn't indicate progress (Reported<br>
by Jan Svoboda)<br>
* ASTERISK-23605 - res_http_websocket: Race condition in shutting<br>
down websocket causes crash (Reported by Matt Jordan)<br>
* ASTERISK-23498 - Asterisk PJSIP transport configuration fails on<br>
parsing of 'cipher' option, any valid option is reported as<br>
unsupported (Reported by Anthony Messina)<br>
* ASTERISK-23672 - PJSIP Digium presence notifications are not<br>
sent if only the subtype or message changes (Reported by Mark<br>
Michelson)<br>
* ASTERISK-23501 - Copy 'Referred-By' header to outgoing INVITE<br>
(Reported by John Bigelow)<br>
* ASTERISK-23707 - Realtime Contacts: Apparent mismatch between<br>
PGSQL database state and Asterisk state (Reported by Mark<br>
Michelson)<br>
* ASTERISK-23675 - [patch] Segmentation Fault on first SIP<br>
registration using res_config_odbc (Reported by Leandro Dardini)<br>
* ASTERISK-23381 - [patch]ChanSpy- Barge only works on the initial<br>
'spy', if the spied-on channel makes a new call, unable to<br>
barge. (Reported by Robert Moss)<br>
* ASTERISK-23497 - chan_sip SIP protocol attended transfer, with<br>
directmedia=yes results in a simple bridge, typically with no<br>
audio (Reported by Etienne Lessard)<br>
* ASTERISK-23665 - Wrong mime type for codec H263-1998 (h263+)<br>
(Reported by Guillaume Maudoux)<br>
* ASTERISK-23664 - Incorrect H264 specification in SDP. (Reported<br>
by Guillaume Maudoux)<br>
* ASTERISK-23709 - Regression in Dahdi/Analog/waitfordialtone<br>
(Reported by Steve Davies)<br>
* ASTERISK-23758 - 500 internal server error when answering a<br>
channel with ARI (Reported by Paul Belanger)<br>
* ASTERISK-22912 - res_corosync doesn't build in Asterisk 12 beta2<br>
(Reported by Malcolm Davenport)<br>
* ASTERISK-22372 - res_corosync: Compilation errors and<br>
functionality broken in Asterisk 12 (Reported by Matt Jordan)<br>
* ASTERISK-23721 - Calls to PJSIP endpoints with video enabled<br>
result in leaked RTP ports (Reported by cervajs)<br>
<br>
New Features made in this release:<br>
-----------------------------------<br>
* ASTERISK-23433 - ARI: Add 'tones' as a URI scheme for /play<br>
operations on resources that support media (bridges, channels)<br>
(Reported by Matt Jordan)<br>
* ASTERISK-22697 - ARI: Add the ability to raise an arbitrary User<br>
Event from the Asterisk or Applications resource (Reported by<br>
Matt Jordan)<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-12.3.0" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.3.0</a><br>
<br>
Thank you for your continued support of Asterisk!<br>
<span class="HOEnZb"><font color="#888888"><br>
--<br>
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