The Asterisk Development Team has announced the first release candidate of Asterisk 12.7.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 12.7.0-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 fixed in this release: ----------------------------------- * ASTERISK-24339 - Swagger API Docs have incorrect basePath (Reported by Bradley Watkins) * ASTERISK-24348 - Built-in editline tab complete segfault with MALLOC_DEBUG (Reported by Walter Doekes) * ASTERISK-24335 - [PATCH] Asterisk incorrectly responds 503 to INVITE retransmissions of rejected calls (Reported by Torrey Searle) * ASTERISK-24295 - crash: creating out of dialog OPTIONS request crashes (Reported by Rogger Padilla) * ASTERISK-23768 - [patch] Asterisk man page contains a (new) unquoted minus sign (Reported by Jeremy Lainé) * ASTERISK-24357 - [fax] Out of bounds error in update_modem_bits (Reported by Jeremy Lainé) * ASTERISK-20567 - bashism in autosupport (Reported by Tzafrir Cohen) * ASTERISK-24350 - PJSIP shows commands prints unneeded headers (Reported by snuffy) * ASTERISK-22945 - [patch] Memory leaks in chan_sip.c with realtime peers (Reported by ibercom) * ASTERISK-24362 - res_hep leaks reference to configuration (Reported by Corey Farrell) * ASTERISK-23781 - outgoing missing as enum from contrib/ast-db-manage/config (Reported by Stephen More) * ASTERISK-24199 - 'ALL' is specified in pjsip.conf.sample for TLS cipher but it is not valid (Reported by Joshua Colp) * ASTERISK-24262 - AMI CoreShowChannel missing several output fields and event documentation (Reported by Mitch Claborn) * ASTERISK-24356 - PJSIP: Directed pickup causes deadlock (Reported by Richard Mudgett) * ASTERISK-24195 - bridge_native_rtp: Removing mixmonitor from a native RTP capable smart bridge doesn't cause the bridge to resume being a native rtp bridge (Reported by Jonathan Rose) * ASTERISK-24384 - chan_motif: format capabilities leak on module load error (Reported by Corey Farrell) * ASTERISK-24385 - chan_sip: process_sdp leaks on an error path (Reported by Corey Farrell) * ASTERISK-24378 - Release AMI connections on shutdown (Reported by Corey Farrell) * ASTERISK-24369 - res_pjsip: Large message on reliable transport can cause empty messages to be passed from the PJSIP stack up, causing crashes in multiple locations (Reported by Matt Jordan) * ASTERISK-24382 - chan_pjsip: Calling PJSIP_MEDIA_OFFER on a non-PJSIP channel results in an invalid reference of a channel pvt and a FRACK (Reported by Matt Jordan) * ASTERISK-24370 - res_pjsip/pjsip_options: OPTIONS request sent to Asterisk with no user in request is always 404'd (Reported by Matt Jordan) * ASTERISK-24224 - When using Bridge() dialplan application, surrogate channel appears in list and call count is inflated. (Reported by Mark Michelson) * ASTERISK-24354 - AMI sendMessage closes AMI connection on error (Reported by Peter Katzmann) * ASTERISK-24398 - Initialize auth_rejection_permanent on client state to the configuration parameter value (Reported by Matt Jordan) * ASTERISK-24326 - res_rtp_asterisk: ICE-TCP candidates are incorrectly attempted (Reported by Joshua Colp) * ASTERISK-24011 - [patch]safe_asterisk tries to set ulimit -n too high on linux systems with lots of RAM (Reported by Michael Myles) * ASTERISK-24383 - res_rtp_asterisk: Crash if no candidates received for component (Reported by Kevin Harwell) * ASTERISK-20784 - Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak (Reported by NITESH BANSAL) * ASTERISK-15879 - [patch] Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak (Reported by Torrey Searle) * ASTERISK-24387 - res_pjsip: rport sent from UAS MUST include the port that the UAC sent the request on (Reported by Matt Jordan) * ASTERISK-24406 - Some caller ID strings are parsed differently since 11.13.0 (Reported by Etienne Lessard) * ASTERISK-24325 - res_calendar_ews: cannot be used with neon 0.30 (Reported by Tzafrir Cohen) * ASTERISK-13797 - [patch] relax badshell tilde test (Reported by Tzafrir Cohen) * ASTERISK-22791 - asterisk sends Re-INVITE after receiving a BYE (Reported by Paolo Compagnini) * ASTERISK-18923 - res_fax_spandsp usage counter is wrong (Reported by Grigoriy Puzankin) * ASTERISK-24394 - CDR: FRACK with PJSIP directed pickup. (Reported by Richard Mudgett) * ASTERISK-24392 - res_fax: fax gateway sessions leak (Reported by Corey Farrell) * ASTERISK-24321 - SIP deadlock when running automated queues tests (Reported by Steve Pitts) * ASTERISK-24393 - rtptimeout=0 doesn't disable rtptimeout (Reported by Dmitry Melekhov) * ASTERISK-23846 - Unistim multilines. Loss of voice after second call drops (on a second line). (Reported by Rustam Khankishyiev) * ASTERISK-24312 - SIGABRT when improperly configured realtime pjsip (Reported by Dafi Ni) * ASTERISK-24426 - CDR Batch mode: size used as time value after first expire (Reported by Shane Blaser) * ASTERISK-24327 - bridge_native_rtp: Smart bridge operation to softmix sometimes fails to properly re-INVITE remotely bridged participants (Reported by Matt Jordan) * ASTERISK-24415 - Missing AMI VarSet events when channels inherit variables. (Reported by Richard Mudgett) * ASTERISK-24063 - [patch]Asterisk does not respect outbound proxy when sending qualify requests (Reported by Damian Ivereigh) * ASTERISK-24122 - Documentaton for res_pjsip option use_avpf needs to be fixed (Reported by James Van Vleet) * ASTERISK-24381 - res_pjsip_sdp_rtp: Declined media streams are interpreted, leading to erroneous 488 rejections (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-24436 - Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0 (Reported by Patrick Laimbock) * ASTERISK-24454 - app_queue: ao2_iterator not destroyed, causing leak (Reported by Corey Farrell) * ASTERISK-24430 - missing letter "p" in word response in OriginateResponse event documentation (Reported by Dafi Ni) * ASTERISK-24437 - Review implementation of ast_bridge_impart for leaks and document proper usage (Reported by Scott Griepentrog) * ASTERISK-24453 - manager: acl_change_sub leaks (Reported by Corey Farrell) * ASTERISK-24457 - res_fax: fax gateway frames leak (Reported by Corey Farrell) * ASTERISK-21721 - SIP Failed to parse multiple Supported: headers (Reported by Olle Johansson) * ASTERISK-24304 - asterisk crashing randomly because of unistim channel (Reported by dhanapathy sathya) * ASTERISK-24190 - IMAP voicemail causes segfault (Reported by Nick Adams) * ASTERISK-24462 - res_pjsip: Stale qualify statistics after disablementation (Reported by Kevin Harwell) * ASTERISK-24466 - app_queue: fix a couple leaks to struct call_queue (Reported by Corey Farrell) * ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled (Reported by Corey Farrell) * ASTERISK-24411 - [patch] Status of outbound registration is not changed upon unregistering. (Reported by John Bigelow) * ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream leaks (Reported by Corey Farrell) For a full list of changes in this release candidate, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.7.0-rc1 Thank you for your continued support of Asterisk!
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