Rilasciato Asterisk 14.3.0
Il giorno 13 febbraio 2017, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 14.3.0.
Dal post originale:
The release of Asterisk 14.3.0 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:
New Features made in this release:
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* ASTERISK-26630 - Make logging PJPROJECT messages a bit easier (Reported by Richard Mudgett)
Bugs fixed in this release:
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* ASTERISK-26772 - Crash in srv.c on startup with pjsip (Reported by nappsoft)
* 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:
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* 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, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.3.0