[asterisk-dev] Asterisk 12.3.0-rc1 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the first release candidate of
Asterisk 12.3.0. This release candidate is available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 12.3.0-rc1 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 candidate:

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)

Improvements 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-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)
 * 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 candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.3.0-rc1

Thank you for your continued support of Asterisk!