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: New Features made in this release: ----------------------------------- * ASTERISK-26277 - Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment (Reported by Matt Jordan) Bugs fixed in this release: ----------------------------------- * ASTERISK-26477 - pjproject: SEGV during SSL operations (Reported by George Joseph) * ASTERISK-17470 - [patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio. (Reported by effie mouzeli) * ASTERISK-26416 - pjproject-bundled: configure fails to check for all required utilities (Reported by Corey Farrell) * ASTERISK-26466 - core: Be forgiving on external callerid that may be flawed so we don't drop events (Reported by Richard Mudgett) * ASTERISK-26362 - res_config_mysql: Broken after 13.10 (Reported by Carlos Chavez) * ASTERISK-26446 - app_dial: There's no way to override the hangupcause on unanswered channels (Reported by George Joseph) * ASTERISK-24311 - Populating database via Alembic fails when using same database for multiple schema sets (Reported by Dafi Ni) * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response. (Reported by Alexander Traud) * ASTERISK-26426 - format_ogg_opus: remove from source (Reported by Kevin Harwell) * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::] (Reported by Jacek) * ASTERISK-25468 - Deadlock in chan_sip - core show locks shows do_monitor lock (Reported by Barry Flanagan) * ASTERISK-26397 - manager: PresenceState action crashes Asterisk 14 (Reported by Andrew Nagy) * ASTERISK-26389 - res_odbc: Clean up pooling options (Reported by Joshua Colp) * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed (Reported by Tzafrir Cohen) * ASTERISK-26273 - core: Won't compile when LOW_MEMORY is enabled (Reported by Anthony Messina) * ASTERISK-26263 - SQL error when using realtime and registering extension / inserting into ps_contacts (Reported by Jeppe Ryskov Larsen) * ASTERISK-26374 - res_pjsip_multihomed: Contact port is rewritten for connectionful protocols (Reported by Joshua Colp) * ASTERISK-26367 - rtp: Timestamps broken when video frame is across multiple RTP packets (Reported by Joshua Colp) * ASTERISK-26375 - res_pjsip_transport_management: Log message states seconds, but time value is milliseconds (Reported by Joshua Colp) * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron Hamstra) * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs. (Reported by Richard Mudgett) * ASTERISK-26316 - res_pjsip_callerid: Irregular URI causes unexpected callerid (Reported by Kevin Harwell) * ASTERISK-26349 - 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed (Reported by Dmitry Melekhov) * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by Etienne Lessard) * ASTERISK-26264 - res_pjsip: Crash when applying ACL from non-existent endpoint (Reported by nappsoft) * ASTERISK-26288 - followme: fails to reset config items to default values on reload (Reported by Tzafrir Cohen) * ASTERISK-25691 - Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up. (Reported by Etienne Lessard) * ASTERISK-26331 - Crash on âcore show channeltype Surrogateâ in ast_format_cap_get_names (Reported by CGI.NET) * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's' extension" (Reported by chris de rock) * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts (Reported by Etienne Lessard) * ASTERISK-26279 - pjproject-bundled: Fails to compile on Debian 6 (Reported by George Joseph) * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up properly (Reported by Alexander Traud) * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members with Local interface (Reported by Etienne Lessard) * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels (Reported by Etienne Lessard) * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code (Reported by David Brillert) * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels. (Reported by Richard Mudgett) * ASTERISK-26269 - res_pjsip: Wrong state for aors without registered contacts after startup (Reported by nappsoft) * ASTERISK-22374 - Finish mapping the sip.conf parameters to res_sip.conf parameters (Reported by Matt Jordan) * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566) (Reported by abelbeck) * ASTERISK-25472 - Swagger scripts are not replacing format variable in file brief (Reported by Corey Farrell) * ASTERISK-26228 - res_pjsip_sdp_rtp: G729A does not include annexb=no attribute. (Reported by Ali Ghavidel) * ASTERISK-25984 - res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it (Reported by József Dudás) * ASTERISK-26305 - Asterisk 14: Two resolver unbound testsuite tests fail (Reported by Richard Mudgett) * ASTERISK-26303 - [patch] BuildSystem: ca_list_path capabilities not detected in PJProject. (Reported by Alexander Traud) * ASTERISK-25492 - ARI: Path parameters are case sensitive (Reported by Joshua Colp) * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names (Reported by Corey Farrell) * ASTERISK-26164 - XMPP no longer triggers NOTIFY to device via chan_pjsip (Reported by Ross Beer) * ASTERISK-26246 - Security: Privilege escalation by AMI adding dialplan extensions. (Reported by Richard Mudgett) * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed startup. (Reported by Corey Farrell) * ASTERISK-26241 - res_pjsip: When using compact headers, rpid and pai are incorrectly generated (Reported by George Joseph) * ASTERISK-26239 - res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname (Reported by George Joseph) * ASTERISK-26238 - res_pjsip: Empty global default_from_user causes crash (Reported by Joshua Colp) * ASTERISK-25797 - app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension (Reported by Etienne Lessard) * ASTERISK-26268 - alembic: 'auth_username' not in PJSIP 'identify_by' enum (Reported by Joshua Colp) * ASTERISK-26145 - pjsip: Deadlock with suspend + masquerade + indicate (Reported by Ross Beer) * ASTERISK-26183 - alembic: error when using sqlalchemy version 1.1.0b2 (Reported by Kevin Harwell) * ASTERISK-26280 - DNS lookups can block channel media paths (Reported by Mark Michelson) * ASTERISK-25217 - [patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c (Reported by Richard Mudgett) * ASTERISK-26265 - Errors ignored from some parts of system initialization. (Reported by Corey Farrell) * ASTERISK-26206 - [patch] res_pjsip: Use more compatible regex for get all (Reported by Dmitry) * ASTERISK-26256 - [patch] SIP/SDP origin (o=) contains brackets with IP6 (Reported by Alexander Traud) * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported by Andrew Nagy) * ASTERISK-26148 - pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..." (Reported by Hans van Eijsden) Improvements made in this release: ----------------------------------- * ASTERISK-26409 - codec_opus: Update Asterisk to support the translation codec. (Reported by Kevin Harwell) * ASTERISK-26289 - Announcer channels in ConfBridges cause inefficiencies (Reported by Mark Michelson) * ASTERISK-25980 - [patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used (Reported by Alexei Gradinari) 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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