The Asterisk Development Team has announced the release of Asterisk 13.12.0.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.12.0 resolves several issues reported by the community and would have not been possible without your participation.
The following are the issues resolved in this release:
- [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
- [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.
- [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, please see the ChangeLog:
Thank you for your continued support of Asterisk!