The Asterisk Development Team has announced the release of Asterisk 12.3.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 12.3.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
Improvements made in this release:
-----------------------------------
* ASTERISK-23553 - Add ast_spinlock capability to lock.h (Reported
by George Joseph)
* ASTERISK-23649 - [patch]Support for DTLS retransmission
(Reported by NITESH BANSAL)
* ASTERISK-23564 - [patch]TLS/SRTP status of channel not currently
available in a CLI command (Reported by Patrick Laimbock)
* ASTERISK-23754 - [patch] Use var/lib directory for log file
configured in asterisk.conf (Reported by Igor Goncharovsky)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-23547 - [patch] app_queue removing callers from queue
when reloading (Reported by Italo Rossi)
* ASTERISK-22846 - testsuite: masquerade super test fails on all
branches (still) (Reported by Matt Jordan)
* ASTERISK-23390 - NewExten Event with application AGI shows up
before and after AGI runs (Reported by Benjamin Keith Ford)
* ASTERISK-23584 - PJSIP 'Unable to create channel' when
attempting to call from endpoint with UDP transport to one using
WebSockets (Reported by Rusty Newton)
* ASTERISK-23545 - Confbridge talker detection settings
configuration load bug (Reported by John Knott)
* ASTERISK-23546 - CB_ADD_LEN does not do what you'd think
(Reported by Walter Doekes)
* ASTERISK-22904 - bridges: lock the bridge when creating bridge
snapshots (Reported by Matt Jordan)
* ASTERISK-23620 - Code path in app_stack fails to unlock list
(Reported by Bradley Watkins)
* ASTERISK-23616 - Big memory leak in logger.c (Reported by
ibercom)
* ASTERISK-23588 - ARI: Crash when unsubscribing from bridge
(Reported by Matt Jordan)
* ASTERISK-23502 - Channel variable SIPREFERTOHDR not being set
during blind transfer (Reported by John Bigelow)
* ASTERISK-23576 - Build failure on SmartOS / Illumos / SunOS
(Reported by Sebastian Wiedenroth)
* ASTERISK-23514 - The pjsip.conf aor qualify contact parameters
are not updated on reload. (Reported by Richard Mudgett)
* ASTERISK-23550 - Newer sound sets don't show up in menuselect
(Reported by Rusty Newton)
* ASTERISK-22677 - Playbacks on bridge via ARI are not queued
(Reported by John Bigelow)
* ASTERISK-18331 - app_sms failure (Reported by David Woodhouse)
* ASTERISK-23487 - features.conf cant load from realtime because
features_config.c starts before loader.c (Reported by Denis)
* ASTERISK-23282 - Documentation - Tab completion and CLI usage
documentation do not indicate that 'all' is accepted for
'confbridge kick all' (Reported by Dorian Logan)
* ASTERISK-19465 - P-Asserted-Identity Privacy (Reported by
Krzysztof Chmielewski)
* ASTERISK-23573 - Crash when transferring unbridged call - in
bridge_app_subscribed at stasis/app.c (Reported by Mark
Michelson)
* ASTERISK-23639 - PJSIP Realtime: Alembic migration needed in
order to widen some string columns (Reported by Mark Michelson)
* ASTERISK-23560 - [ARI] MOH doesn't indicate progress (Reported
by Jan Svoboda)
* ASTERISK-23605 - res_http_websocket: Race condition in shutting
down websocket causes crash (Reported by Matt Jordan)
* ASTERISK-23498 - Asterisk PJSIP transport configuration fails on
parsing of 'cipher' option, any valid option is reported as
unsupported (Reported by Anthony Messina)
* ASTERISK-23672 - PJSIP Digium presence notifications are not
sent if only the subtype or message changes (Reported by Mark
Michelson)
* ASTERISK-23501 - Copy 'Referred-By' header to outgoing INVITE
(Reported by John Bigelow)
* ASTERISK-23707 - Realtime Contacts: Apparent mismatch between
PGSQL database state and Asterisk state (Reported by Mark
Michelson)
* ASTERISK-23675 - [patch] Segmentation Fault on first SIP
registration using res_config_odbc (Reported by Leandro Dardini)
* ASTERISK-23381 - [patch]ChanSpy- Barge only works on the initial
'spy', if the spied-on channel makes a new call, unable to
barge. (Reported by Robert Moss)
* ASTERISK-23497 - chan_sip SIP protocol attended transfer, with
directmedia=yes results in a simple bridge, typically with no
audio (Reported by Etienne Lessard)
* ASTERISK-23665 - Wrong mime type for codec H263-1998 (h263+)
(Reported by Guillaume Maudoux)
* ASTERISK-23664 - Incorrect H264 specification in SDP. (Reported
by Guillaume Maudoux)
* ASTERISK-23709 - Regression in Dahdi/Analog/waitfordialtone
(Reported by Steve Davies)
* ASTERISK-23758 - 500 internal server error when answering a
channel with ARI (Reported by Paul Belanger)
* ASTERISK-22912 - res_corosync doesn't build in Asterisk 12 beta2
(Reported by Malcolm Davenport)
* ASTERISK-22372 - res_corosync: Compilation errors and
functionality broken in Asterisk 12 (Reported by Matt Jordan)
* ASTERISK-23721 - Calls to PJSIP endpoints with video enabled
result in leaked RTP ports (Reported by cervajs)
New Features made in this release:
-----------------------------------
* ASTERISK-23433 - ARI: Add 'tones' as a URI scheme for /play
operations on resources that support media (bridges, channels)
(Reported by Matt Jordan)
* ASTERISK-22697 - ARI: Add the ability to raise an arbitrary User
Event from the Asterisk or Applications resource (Reported by
Matt Jordan)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.3.0
Thank you for your continued support of Asterisk!
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