The Asterisk Development Team has announced the first release candidate of Asterisk 14.3.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 14.3.0-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-26630 - Make logging PJPROJECT messages a bit easier (Reported by Richard Mudgett) Bugs fixed in this release: ----------------------------------- * ASTERISK-26767 - ARI channelvars cause memory leak (Reported by Sébastien Duthil) * ASTERISK-26716 - ari: Channels with pre-dial handlers cannot be hung up via ARI (Reported by Tom Pawelek) * ASTERISK-26632 - core: Possibility of a frame "imbalance" leading to stuck channels. (Reported by Mark Michelson) * ASTERISK-25951 - res_agi: run_agi eats frames it shouldn't (Reported by George Joseph) * ASTERISK-26343 - ASTERISK-25951 causes issues for callerid manipulation through agi (Reported by Morten Tryfoss) * ASTERISK-26704 - res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'. (Reported by Anthony Messina) * ASTERISK-26765 - res_resolver_unbound: FRACK! Excessive ref count trap tripped. (Reported by Richard Mudgett) * ASTERISK-21094 - MixMonitorMute mutes through stream if already slinear (e.g. Originate) (Reported by David Woolley) * ASTERISK-26679 - Crash on invalid contact domain (pjsip aor) (Reported by Dmitriy) * ASTERISK-26699 - res_pjsip: Assertion when sending OPTIONS request to endpoint (Reported by Ross Beer) * ASTERISK-24858 - [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec (Reported by Frankie Chin) * ASTERISK-26754 - build_tools: make_build_h does not handle \ in user name (Reported by Kirill Katsnelson) * ASTERISK-26753 - AMI disconnect causes "ast_careful_fwrite: fwrite() returned error: Broken pipe" (Reported by Kirill Katsnelson) * ASTERISK-26755 - app_queue: Random queues disappear on "core reload queue all" (Reported by Kirill Katsnelson) * ASTERISK-26735 - res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect (Reported by Michael Maier) * ASTERISK-26693 - res_pjsip_endpoint_identifier_ip: Add support for SRV (Reported by Joshua Colp) * ASTERISK-26743 - PJPROJECT: Detecting compiled max log level does not work. (Reported by Richard Mudgett) * ASTERISK-26740 - voicemail API test: uses varlibdir instead of datadir for a sound file (Reported by Tzafrir Cohen) * ASTERISK-26739 - voicemail API test: confuses expected and actual values (Reported by Tzafrir Cohen) * ASTERISK-26731 - res_sorcery_memory_cache: memory leak on every sorcery memory cache populate (Reported by Ustinov Artem) * ASTERISK-26710 - [patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0 (Reported by Aaron An) * ASTERISK-26670 - [patch] Outgoing SIP-URI Dialing via PJSIP (Reported by Alexander Traud) * ASTERISK-26691 - Remember SDP negotiation on SIP_CODEC_INBOUND. (Reported by Alexander Traud) * ASTERISK-26673 - chan_pjsip: Crash when using CHANNEL dialplan function around masquerade (Reported by Joshua Colp) * ASTERISK-26684 - res_pjsip: Various issues with compact SIP headers (Reported by Joshua Elson) * ASTERISK-26655 - [patch]pjsip: Transfers Broken with Compact Headers Enabled (Reported by JoshE) * ASTERISK-26672 - Crash when setting remote address on RTP instance (Reported by Richard Mudgett) * ASTERISK-26621 - app_queue: Queue application does not ring members with Local interface (Reported by Jonas Kellens) * ASTERISK-26586 - chan_sip: Segfaults upon reload if client with MWI wasn't registered (Reported by Michael Kuron) * ASTERISK-25494 - build: GCC 5.1.x catches some new const, array bounds and missing paren issues (Reported by George Joseph) * ASTERISK-24499 - Need more explicit debug when PJSIP dialstring is invalid (Reported by Rusty Newton) * ASTERISK-25083 - Message.c: Message channel becomes saturated with frames leading to spammy log messages (Reported by Jonathan Rose) * ASTERISK-26653 - pjproject_bundled doesn't verify already downloaded tarballs (Reported by George Joseph) * ASTERISK-26433 - chan_sip: Allows To-tag checks to be bypassed, setting up new calls (Reported by Walter Doekes) * ASTERISK-26579 - codec_opus: Recursiveness when parsing fmtp line (Reported by Jørgen H) * ASTERISK-26644 - PJSIPShowRegistrationsInbound just dumps all aors (Reported by George Joseph) * ASTERISK-26647 - Support older DNS style for OpenBSD (Reported by snuffy) * ASTERISK-26490 - res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_" (Reported by Juris Breicis) * ASTERISK-26617 - res_rtp_asterisk: Can't bind on systems without IPv6 (Reported by Guido Falsi) * ASTERISK-26603 - [patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no (Reported by Alexei Gradinari) * ASTERISK-24330 - Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118 (Reported by Marek Cervenka) * ASTERISK-26546 - mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2' (Reported by Tzafrir Cohen) * ASTERISK-26566 - res_rtp_asterisk: RTT miscalculation in RTCP (Reported by Hector Royo Concepcion) * ASTERISK-26604 - chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc. (Reported by Michael Kuron) * ASTERISK-26608 - Compile and link failures on OpenBSD (Reported by snuffy) Improvements made in this release: ----------------------------------- * ASTERISK-23828 - pjsip - Need a command to list active SIP subscriptions (Reported by Rusty Newton) * ASTERISK-26527 - Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec (Reported by Badalian Vyacheslav) * ASTERISK-26624 - res_calendar_caldav: Add support for gmail (Reported by Eduardo Scudeller Libardi) * ASTERISK-26562 - app_controlplayback: Transmit Silence on ControlPlayback pause (Reported by Mikheili Dautashvili) For a full list of changes in this release candidate, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.3.0-rc1 Thank you for your continued support of Asterisk!
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