Asterisk 13.12.0-rc1 Now Available

The Asterisk Development Team has announced the first release candidate of Asterisk 13.12.0.

This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.12.0-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-17470] – – When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
  • [ASTERISK-18232] – Broken REGISTER sent to IPv4 server when bindaddr=[::]
  • [ASTERISK-19968] – TCP Session-Timers not dropping call
  • [ASTERISK-22374] – Finish mapping the sip.conf parameters to res_sip.conf parameters
  • [ASTERISK-22732] – Deadlock potential in res_fax and CCSS with local channels.
  • [ASTERISK-24311] – Populating database via Alembic fails when using same database for multiple schema sets
  • [ASTERISK-24425] – jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
  • [ASTERISK-24822] – Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
  • [ASTERISK-25217] – res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
  • [ASTERISK-25468] – Deadlock in chan_sip – core show locks shows do_monitor lock
  • [ASTERISK-25472] – Swagger scripts are not replacing format variable in file brief
  • [ASTERISK-25492] – ARI: Path parameters are case sensitive
  • [ASTERISK-25691] – Crash occurs when screening mode (Dial’s ‘p’ argument) is enabled and callee rejects a call or hangs up.
  • [ASTERISK-25797] – app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
  • [ASTERISK-25984] – res_odbc relies on res_odbc_transaction, but it’s not mandatory to compile it
  • [ASTERISK-25996] – Remove “live_dangerously” requirement on DB(read)
  • [ASTERISK-26145] – pjsip: Deadlock with suspend + masquerade + indicate
  • [ASTERISK-26148] – pjsip: Cannot compile 13.10.0-rc1: “libasteriskpj.so: undefined reference to…”
  • [ASTERISK-26164] – XMPP no longer triggers NOTIFY to device via chan_pjsip
  • [ASTERISK-26183] – alembic: error when using sqlalchemy version 1.1.0b2
  • [ASTERISK-26203] – res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
  • [ASTERISK-26206] – res_pjsip: Use more compatible regex for get all
  • [ASTERISK-26226] – pbx: Asterisk crash on AMI action “ShowDialplan” when there’s a circular dependency between contexts
  • [ASTERISK-26228] – res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
  • [ASTERISK-26233] – pbx: Failure to remove inconsistent extension names
  • [ASTERISK-26238] – res_pjsip: Empty global default_from_user causes crash
  • [ASTERISK-26239] – res_pjsip_logger: An empty global/debug option is treated as a “match all” hostname
  • [ASTERISK-26241] – res_pjsip: When using compact headers, rpid and pai are incorrectly generated
  • [ASTERISK-26246] – Security: Privilege escalation by AMI adding dialplan extensions.
  • [ASTERISK-26256] – SIP/SDP origin (o=) contains brackets with IP6
  • [ASTERISK-26263] – SQL error when using realtime and registering extension / inserting into ps_contacts
  • [ASTERISK-26264] – res_pjsip: Crash when applying ACL from non-existent endpoint
  • [ASTERISK-26265] – Errors ignored from some parts of system initialization.
  • [ASTERISK-26267] – ast_register_atexit callbacks should be run on failed startup.
  • [ASTERISK-26268] – alembic: ‘auth_username’ not in PJSIP ‘identify_by’ enum
  • [ASTERISK-26269] – res_pjsip: Wrong state for aors without registered contacts after startup
  • [ASTERISK-26272] – chan_sip: File descriptors leak (UDP sockets)
  • [ASTERISK-26273] – core: Won’t compile when LOW_MEMORY is enabled
  • [ASTERISK-26279] – pjproject-bundled: Fails to compile on Debian 6
  • [ASTERISK-26280] – DNS lookups can block channel media paths
  • [ASTERISK-26282] – AEL: macro-call in Dial application, macro “lacks ‘s’ extension”
  • [ASTERISK-26288] – followme: fails to reset config items to default values on reload
  • [ASTERISK-26299] – app_queue: Queue application sometimes stops calling members with Local interface
  • [ASTERISK-26303] – BuildSystem: ca_list_path capabilities not detected in PJProject.
  • [ASTERISK-26305] – Asterisk 14: Two resolver unbound testsuite tests fail
  • [ASTERISK-26306] – channel: Hang-up crashes, chan_pjsip not cleaning up properly
  • [ASTERISK-26316] – res_pjsip_callerid: Irregular URI causes unexpected callerid
  • [ASTERISK-26331] – Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
  • [ASTERISK-26349] – 13.11.1 res_pjsip/pjsip_distributor.c: Request ‘REGISTER’ failed
  • [ASTERISK-26359] – cdr_mysql: fails to use UTC if so instructed
  • [ASTERISK-26360] – app_queue: “queue show” output gets “failed to extend from 240 to 327” msgs.
  • [ASTERISK-26362] – res_config_mysql: Broken after 13.10
  • [ASTERISK-26367] – rtp: Timestamps broken when video frame is across multiple RTP packets
  • [ASTERISK-26374] – res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
  • [ASTERISK-26375] – res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
  • [ASTERISK-26389] – res_odbc: Clean up pooling options
  • [ASTERISK-26397] – manager: PresenceState action crashes Asterisk 14
  • [ASTERISK-26416] – pjproject-bundled: configure fails to check for all required utilities
  • [ASTERISK-26426] – format_ogg_opus: remove from source
  • [ASTERISK-26438] – chan_sip: auto_force_rport: No NAT = No Symmetric Response.
  • [ASTERISK-26446] – app_dial: There’s no way to override the hangupcause on unanswered channels
  • [ASTERISK-26466] – core: Be forgiving on external callerid that may be flawed so we don’t drop events
  • [ASTERISK-26477] – pjproject: SEGV during SSL operations

Improvement

  • [ASTERISK-25980] – res_fax: set FAXMODE variable to let dialplan know what fax transport was used
  • [ASTERISK-26289] – Announcer channels in ConfBridges cause inefficiencies
  • [ASTERISK-26409] – codec_opus: Update Asterisk to support the translation codec.

New Feature

  • [ASTERISK-26277] – Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.12.0-rc1

Thank you for your continued support of Asterisk!

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