Asterisk 12.3.0 Now Available

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:

Bug

  • [ASTERISK-18331] – app_sms failure
  • [ASTERISK-19465] – P-Asserted-Identity Privacy
  • [ASTERISK-22372] – res_corosync: Compilation errors and functionality broken in Asterisk 12
  • [ASTERISK-22677] – Playbacks on bridge via ARI are not queued
  • [ASTERISK-22846] – testsuite: masquerade super test fails on all branches (still)
  • [ASTERISK-22904] – bridges: lock the bridge when creating bridge snapshots
  • [ASTERISK-22912] – res_corosync doesn’t build in Asterisk 12 beta2
  • [ASTERISK-23282] – Documentation – Tab completion and CLI usage documentation do not indicate that ‘all’ is accepted for ‘confbridge kick all’
  • [ASTERISK-23381] – ChanSpy- Barge only works on the initial ‘spy’, if the spied-on channel makes a new call, unable to barge.
  • [ASTERISK-23390] – NewExten Event with application AGI shows up before and after AGI runs
  • [ASTERISK-23487] – features.conf cant load from realtime because features_config.c starts before loader.c
  • [ASTERISK-23497] – chan_sip SIP protocol attended transfer, with directmedia=yes results in a simple bridge, typically with no audio
  • [ASTERISK-23498] – Asterisk PJSIP transport configuration fails on parsing of ‘cipher’ option, any valid option is reported as unsupported
  • [ASTERISK-23501] – Copy ‘Referred-By’ header to outgoing INVITE
  • [ASTERISK-23502] – Channel variable SIPREFERTOHDR not being set during blind transfer
  • [ASTERISK-23514] – The pjsip.conf aor qualify contact parameters are not updated on reload.
  • [ASTERISK-23545] – Confbridge talker detection settings configuration load bug
  • [ASTERISK-23546] – CB_ADD_LEN does not do what you’d think
  • [ASTERISK-23547] – app_queue removing callers from queue when reloading
  • [ASTERISK-23550] – Newer sound sets don’t show up in menuselect
  • [ASTERISK-23560] – [ARI] MOH doesn’t indicate progress
  • [ASTERISK-23573] – Crash when transferring unbridged call – in bridge_app_subscribed at stasis/app.c
  • [ASTERISK-23576] – Build failure on SmartOS / Illumos / SunOS
  • [ASTERISK-23584] – PJSIP ‘Unable to create channel’ when attempting to call from endpoint with UDP transport to one using WebSockets
  • [ASTERISK-23588] – ARI: Crash when unsubscribing from bridge
  • [ASTERISK-23605] – res_http_websocket: Race condition in shutting down websocket causes crash
  • [ASTERISK-23616] – Big memory leak in logger.c
  • [ASTERISK-23620] – Code path in app_stack fails to unlock list
  • [ASTERISK-23639] – PJSIP Realtime: Alembic migration needed in order to widen some string columns
  • [ASTERISK-23664] – Incorrect H264 specification in SDP.
  • [ASTERISK-23665] – Wrong mime type for codec H263-1998 (h263+)
  • [ASTERISK-23672] – PJSIP Digium presence notifications are not sent if only the subtype or message changes
  • [ASTERISK-23675] – Segmentation Fault on first SIP registration using res_config_odbc
  • [ASTERISK-23707] – Realtime Contacts: Apparent mismatch between PGSQL database state and Asterisk state
  • [ASTERISK-23709] – Regression in Dahdi/Analog/waitfordialtone
  • [ASTERISK-23721] – Calls to PJSIP endpoints with video enabled result in leaked RTP ports
  • [ASTERISK-23758] – 500 internal server error when answering a channel with ARI

Improvement

  • [ASTERISK-23553] – Add ast_spinlock capability to lock.h
  • [ASTERISK-23564] – TLS/SRTP status of channel not currently available in a CLI command
  • [ASTERISK-23649] – Support for DTLS retransmission
  • [ASTERISK-23754] – Use var/lib directory for log file configured in asterisk.conf

New Feature

  • [ASTERISK-22697] – ARI: Add the ability to raise an arbitrary User Event from the Asterisk or Applications resource
  • [ASTERISK-23433] – ARI: Add ‘tones’ as a URI scheme for /play operations on resources that support media (bridges, channels)

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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