Asterisk 12.1.0-rc1 Now Available

The Asterisk Development Team has announced the first release candidate of Asterisk 12.1.0. This release candidate is available for immediate download at  http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 12.1.0-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

  • [ASTERISK-17138] – Asterisk not re-registering after it receives “Forbidden – wrong password on authentication”
  • [ASTERISK-17727] – TLS doesn’t get all certificate chain
  • [ASTERISK-17837] – extconfig.conf – Maximum Include level (1) exceeded
  • [ASTERISK-19773] – Asterisk crash on issuing Asterisk-CLI ‘reload’ command multiple times on cli_aliases
  • [ASTERISK-22486] – ARI: TCP Reset after 204 response
  • [ASTERISK-22662] – Documentation fix? – queues.conf says persistentmembers defaults to yes, it appears to lie
  • [ASTERISK-22757] – segfault in res_clialiases.so on reload when mapping “module reload” command
  • [ASTERISK-22790] – check_modem_rate() may return incorrect rate for V.27
  • [ASTERISK-22854] – – Deadlock between cel_pgsql unload and core_event_dispatcher taskprocessor thread
  • [ASTERISK-22861] – Specifying a null time as parameter to GotoIfTime or ExecIfTime causes segmentation fault
  • [ASTERISK-22871] – cel_pgsql module not loading after “reload” or “reload cel_pgsql.so” command
  • [ASTERISK-22884] – hangup_handler end with h extension: tests currently fail in Asterisk 12 +
  • [ASTERISK-22910] – – REPLACE() calls strcpy on overlapping memory when <replace-char> is empty
  • [ASTERISK-22911] – Asterisk fails to resume WebRTC call from hold
  • [ASTERISK-22924] – PJSIP MESSAGE support does not present the contact information on outbound messages
  • [ASTERISK-22946] – Local From tag regression with sipgate.de
  • [ASTERISK-22952] – res_pjsip_pubsub: crash when subscription_destructor is terminated from a non-PJSIP thread
  • [ASTERISK-22962] – performance spike on Local channels originated using ARI
  • [ASTERISK-22988] – T38 , SIP 488 after Rejecting image media offer due to invalid or unsupported syntax
  • [ASTERISK-23008] – Local channels loose CALLERID name when DAHDI channel connects
  • [ASTERISK-23011] – configure.ac and pbx_lua don’t support lua 5.2
  • [ASTERISK-23018] – PJSip ‘allow=all’ results in failed SDP negotiation
  • [ASTERISK-23027] – Spelling typo “transfered” instead of “transferred”
  • [ASTERISK-23028] – Asterisk man pages contains unquoted minus signs
  • [ASTERISK-23034] – manager Originate doesn’t abort on failed format_cap allocation
  • [ASTERISK-23046] – Custom CDR fields set during a GoSUB called from app_queue are not inserted
  • [ASTERISK-23051] – ARI: channel variables in JSON breaks passing parameters in JSON
  • [ASTERISK-23053] – The users of ao2_iterator_cleanup() are violating the ao2_iterator opacity.
  • [ASTERISK-23056] – INFINITY and NAN undefined
  • [ASTERISK-23061] – [Patch] ‘textsupport’ setting not mentioned in sip.conf.sample
  • [ASTERISK-23062] – res_pjsip AOR config option qualify_frequency is inconsistently respected
  • [ASTERISK-23065] – On Asterisk start, device state is INVALID for previously registered PJSIP endpoints, despite re-registrations
  • [ASTERISK-23071] – pjsip: mailboxes documentation is lacking
  • [ASTERISK-23072] – MWI subscription from Cisco SPA fails with PJSIP
  • [ASTERISK-23074] – Crash in ChanIsAvail app
  • [ASTERISK-23081] – PJSip Tab Expansion erroring
  • [ASTERISK-23082] – Including g722 in pjsip codec configuration results in unexpected SDP offers
  • [ASTERISK-23084] – rasterisk needlessly prints the AST-2013-007 warning
  • [ASTERISK-23100] – In chan_mgcp the ident in transmitted request and request queue may differ – fix for locking
  • [ASTERISK-23101] – pjsip: crash when parsing scheme from SIP URI
  • [ASTERISK-23106] – pjsip: ACK to 200 OK sent to private IP address on outbound channel’s INVITE request
  • [ASTERISK-23128] – res_ari: Memory leak on response headers and some JSON response messages
  • [ASTERISK-23129] – segfault in res_pjsip_pubsub.so
  • [ASTERISK-23134] – res_rtp_asterisk port selection cannot handle selinux port restrictions
  • [ASTERISK-23143] – ARI: subscribing to an already subscribed resource returns a 500 error
  • [ASTERISK-23164] – CDRs: mid-call/pre-dial handlers perturb context/exten/app/data fields during Dial
  • [ASTERISK-23168] – Overriding outbound_auth in a pjsip registration causes ERROR, assert failure.
  • [ASTERISK-23177] – RealTime cant update sipbuddies table when registering or updating friend
  • [ASTERISK-23178] – devicestate.h: device state setting functions are documented with the wrong return values
  • [ASTERISK-23220] – STACK_PEEK function with no arguments causes crash/core dump
  • [ASTERISK-23249] – Skinny subchannel locking issues
  • [ASTERISK-23250] – CDR(start) function is broken due to sizeof dereference

Improvement

  • [ASTERISK-21084] – New SIP Channel Driver – Path Support
  • [ASTERISK-22610] – Implement CLI commands for PJSIP
  • [ASTERISK-22868] – chan_pjsip: ‘setvar’ should be supported on endpoints
  • [ASTERISK-22918] – dahdi show channels slices PRI channel dnid on output
  • [ASTERISK-22919] – core show channeltypes slicing
  • [ASTERISK-22980] – Allow building cdr_radius and cel_radius against libfreeradius-client
  • [ASTERISK-22984] – ari: Transfer messages not being sent out ARI WebSocket
  • [ASTERISK-23068] – http: Implement support for chunked Transfer-Encoding

New Feature

  • [ASTERISK-23038] – Need config option to enable PJSIP logger at load time

For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/Asterisk/ChangeLog-12.1.0-rc1

Thank you for your continued support of Asterisk!

Scroll to Top