The Asterisk Development Team would like to announce the first release candidate of Asterisk 15.2.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.2.0-rc1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release candidate: New Features made in this release: ----------------------------------- * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming INVITE Request-URI. (Reported by Richard Mudgett) * ASTERISK-27413 - Add cache_media_frames debugging option. (Reported by Richard Mudgett) * ASTERISK-27206 - res_pjsip: No mechanism exists to limit endpoint identification to IP only (Reported by Ben Merrills) Bugs fixed in this release: ----------------------------------- * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on read() (Reported by Abhay Gupta) * ASTERISK-25079 - AMI bridge of channels results in MOH not destroyed and robotic audio on one channel (Reported by Zane Conkle) * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash (Reported by Corey Farrell) * ASTERISK-27490 - chan_console: 'set active' fails to work (Reported by Tzafrir Cohen) * ASTERISK-24756 - ConfBridge sound_muted does not work from CLI or AMI (Reported by Thomas Frederiksen) * ASTERISK-25649 - Transfer application does not work with Local channels - documentation misleading (Reported by Ivan Ullmann) * ASTERISK-25869 - chan_sip: "rejected because extension not found" should be logged as a security event (Reported by Brian J. Murrell) * ASTERISK-27440 - Strictrtp has issues to qualify video rtp streams (Reported by Wim De Vlaminck) * ASTERISK-24329 - Music On Hold announcement cuts intro of music the first time it is played (Reported by Thomas Frederiksen) * ASTERISK-19657 - Coverity Report: Fix issues for error type CHAR_IO (Reported by Matt Jordan) * ASTERISK-27175 - iax.conf demo peer is invalid (Reported by Tzafrir Cohen) * ASTERISK-27430 - README refers to security documents that do not exist. (Reported by Corey Farrell) * ASTERISK-20281 - "core set verbose" behaves strangely, can't alias it, cli.conf example broken (Reported by Tim Ringenbach at Asteria Solutions Group) * ASTERISK-27382 - crash after an invalid rtcp packet from GT48 FXS gateway (Reported by Tzafrir Cohen) * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an RTCP packet will write past where it should (Reported by Vitezslav Novy) * ASTERISK-27408 - Identify causes and fix pjsip/resolver/srv/failover/in_dialog/transport_tcp (Reported by Corey Farrell) * ASTERISK-18411 - Queue members with hints for state_interface get stuck in "In Use" state. (Reported by Steven T. Wheeler) * ASTERISK-26131 - chan_sip: Crash Asterisk (in sip_request_call at chan_sip.c) by making a call to a single character in a dot pattern match (Reported by Dwayne Hubbard) * ASTERISK-27475 - codec_opus requires libcurl (Reported by Samuel For) * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes not applied on reload (Reported by John Bigelow) * ASTERISK-27465 - CLI Completion Not Working (Reported by Ross Beer) * ASTERISK-27460 - CDR: Deadlock using AMI Originate with Variable CDR(amaflags)=... (Reported by Richard Mudgett) * ASTERISK-27453 - RTP: Blind transfer direct media scenario results in one way audio. (Reported by Richard Mudgett) * ASTERISK-20643 - SIP ICE support - remove hardcoded limitation on SDP size, make ICE support disabled by default in SIP, maybe provide a better warning message (Reported by Roy) * ASTERISK-26980 - pjsip: Clean up WebRTC disables (Reported by abelbeck) * ASTERISK-27452 - Security: chan_skinny: Memory exhaustion if flooded with unauthenticated requests (Reported by George Joseph) * ASTERISK-27454 - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported by Joshua Colp) * ASTERISK-23735 - Transcoding makes bad choice in high-rate translations (Reported by Richard Kenner) * ASTERISK-27445 - ARI: Updating a bridge gives wrong error message. (Reported by Frank Durden) * ASTERISK-24662 - [patch] column and row headers for Signed Linear format variants in output of 'core show translation' are ambiguous (Reported by Rusty Newton) * ASTERISK-27353 - H323 audio starts with a delay of 2 seconds. (Reported by Marco Giordani) * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate media (Reported by Kevin Harwell) * ASTERISK-27437 - [patch] ICE: server-reflexive candidates (srflx) with Dual-Stack. (Reported by Alexander Traud) * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around IPv6 addresses. (Reported by Alexander Traud) * ASTERISK-27435 - [patch] configure: pjsip_evsub_set_uas_timeout not found. (Reported by Alexander Traud) * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra (Reported by Ivan Larionov) * ASTERISK-27431 - Asterisk fails to build when openssl headers are not installed. (Reported by Corey Farrell) * ASTERISK-27421 - RTP source learning not working with devices that have some clock issues (Reported by nappsoft) * ASTERISK-27361 - Attended transfer crashes in Asterisk 13.17.2 (Reported by Alessandro Pimenta) * ASTERISK-27238 - Bridging: Crash freeing a frame that's already been freed (Reported by Richard Kenner) * ASTERISK-27412 - core: Audiohook freeing interpolated frame when it shouldn't. (Reported by Mikhail) * ASTERISK-27423 - app_record: We set the RECORD_STATUS channel variable before closing the file (Reported by George Joseph) * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk insert same ip address in "source ip address" and "destination ip address" fields in HEP packets (Reported by Max Norba) * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it is equal to RemoteAddress) (Reported by Vasilii Rogin) * ASTERISK-27415 - asterisk.conf: Setting astctl without setting astrundir is ineffective. (Reported by Corey Farrell) * ASTERISK-27411 - pjsip: TCP connections may not be destroyed (Reported by Joshua Colp) * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488 responses. (Reported by Corey Farrell) * ASTERISK-27337 - chan_sip: Security vulnerability with client code header (revisited) (Reported by Richard Mudgett) * ASTERISK-27319 - (Security) Function in PJSIP 2.7 miscalculates the length of an unsigned long variable in 64bit machines (Reported by Kim youngsung) * ASTERISK-27391 - Regression: Deadlock between AOR named lock and pjproject grp lock (Reported by shaurya jain) * ASTERISK-27393 - res_pjsip: Crash occurs when an empty contact read from astdb or database (Reported by Aaron An) * ASTERISK-27290 - res_pjsip: PIDF contact field has malformed/invalid XML (Reported by basildane) * ASTERISK-27032 - res_pjsip: TLS options do not handle empty values (Reported by seanchann.zhou) * ASTERISK-27395 - srtp: Add support for ephemeral DTLS certificates (Reported by Sean Bright) * ASTERISK-26426 - format_ogg_opus: remove from source (Reported by Kevin Harwell) * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is enabled but not configured. (Reported by Alexander Traud) * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support (Reported by Alexander Traud) * ASTERISK-27378 - Modules: Fix issues with CLI completion. (Reported by Corey Farrell) * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+" character isn't allowed any more (Reported by Michael Maier) * ASTERISK-27364 - channel: Crash when fax gateway is in use with PJSIP (Reported by Jared Hull) * ASTERISK-27390 - Audit menuselect module dependencies (Reported by Corey Farrell) * ASTERISK-27389 - Optional API modules should not allow unload. (Reported by Corey Farrell) * ASTERISK-27369 - Bridge() dialplan application fails without setting BRIDGERESULT channel variable (Reported by James Terhune) * ASTERISK-27067 - res_ari_channels: channel_state_invalid always leaks snapshot reference. (Reported by Marin Odrljin) * ASTERISK-27379 - stream: Allow streams on a topology to be put into groups (Reported by Joshua Colp) * ASTERISK-27374 - alembic: PJSIP scripts are missing column bundle in ps_endpoints table (Reported by Florian Floimair) * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage documentation (Reported by Igor Goncharovsky) * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function 'imap_delete_old_greeting' (Reported by Anthony Messina) * ASTERISK-27194 - jitterbuffer: Does not handle case where translator returns null frame. (Reported by Joshua Elson) * ASTERISK-27372 - ARI: Node ARI client broken in latest versions of 13 and 14 (Reported by Benjamin Keith Ford) * ASTERISK-26639 - core: Disabling xmldoc support does not work. Also results in abort during Asterisk startup. (Reported by Mr Dini) * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the absence of the Expires header field with an unsubscribe action. (Reported by Jonathan Cloots) * ASTERISK-25960 - The config_hook unit test causes Asterisk to crash if run a second time (Reported by George Joseph) * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6 set to yes (Reported by Martin Cisárik) * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded (Reported by Ronald Raikes) * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last but first on SDP media level. (Reported by Alexander Traud) * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on un/re-load: mod.id == -1 (Reported by Tzafrir Cohen) * ASTERISK-23462 - Cannot disable SIP debugging via CLI after enabling with conf file option - also 'sip set debug off' reports debugging disabled, when it really isn't (Reported by Rusty Newton) * ASTERISK-27354 - bridge_softmix: When a channel leaves add in any missing participant streams (Reported by Joshua Colp) * ASTERISK-27333 - sip_to_pjsip not correctly handling disallow=all directive (Reported by Torrey Searle) * ASTERISK-27328 - Missing openssl dependencies in res_rtp_asterisk and tcptls (Reported by Tzafrir Cohen) * ASTERISK-27343 - Fails to build in FreeBSD due to sys/sysmacros.h not existing there (Reported by Guido Falsi) * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin (o=) contains local address. (Reported by Alexander Traud) * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller (Reported by lvl) * ASTERISK-27340 - backtrace.c: Crash due to double-free. (Reported by Corey Farrell) * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when stopping. (Reported by Alexander Traud) * ASTERISK-27416 - Can't load res_corosync.so module on Asterisk 13.18.2 (Reported by Anton Mosin) Improvements made in this release: ----------------------------------- * ASTERISK-24297 - cdr.c: Minor code optimizations. (Reported by Richard Mudgett) * ASTERISK-27449 - [PATCH] When failing to acquire target during attended transfer, display wanted extension (Reported by Niklas Larsson) * ASTERISK-27456 - app_voicemail: Add new object for VoicemailUserEntry (Reported by sungtae kim) * ASTERISK-27380 - ast_coredumper: allow pointing out the asterisk binary explicitly (Reported by Tzafrir Cohen) * ASTERISK-23556 - Compilation warning for invert.c (array subscript is above array bounds) (Reported by Marcello Ceschia) * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard Mudgett) * ASTERISK-27335 - CDR performance needs improvement. (Reported by Richard Mudgett) For a full list of changes in this release candidate, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0-rc1 Thank you for your continued support of Asterisk!