[asterisk-dev] Asterisk 12.3.0 Now Available

[Asterisk Development Team]

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!