Asterisk 12.1.0 Now Available

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

The release of Asterisk 12.1.0 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:

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-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-23213] – SIP over WS: Audio problems when upgrading to 11.8 from 11.7 with endpoints behind NAT
  • [ASTERISK-23220] – STACK_PEEK function with no arguments causes crash/core dump
  • [ASTERISK-23231] – Since 405693 If we have res_fax.conf file set to minrate=2400, then res_fax refuse to load
  • [ASTERISK-23249] – Skinny subchannel locking issues
  • [ASTERISK-23250] – CDR(start) function is broken due to sizeof dereference

Improvements

  • [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 Features

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

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

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

Thank you for your continued support of Asterisk!

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