Asterisk 12.3.0-rc1 Now Available

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:

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-23758] – 500 internal server error when answering a channel with ARI

Improvement

  • [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)
  • [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

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!

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