The Asterisk Development Team would like to announce the release of Asterisk 13.19.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.19.0 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: *New Features made in this release:* ----------------------------------- - [ASTERISK-27478 <https://issues.asterisk.org/jira/browse/ASTERISK-27478>] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming INVITE Request-URI. (Reported by Richard Mudgett) - [ASTERISK-27413 <https://issues.asterisk.org/jira/browse/ASTERISK-27413>] - Add cache_media_frames debugging option. (Reported by Richard Mudgett) - [ASTERISK-27206 <https://issues.asterisk.org/jira/browse/ASTERISK-27206>] - res_pjsip: No mechanism exists to limit endpoint identification to IP only (Reported by Ben Merrills) *Bugs fixed in this release:* ----------------------------------- - [ASTERISK-27531 <https://issues.asterisk.org/jira/browse/ASTERISK-27531>] - Compiler optimizations can break module load sequence. (Reported by abelbeck) - [ASTERISK-27480 <https://issues.asterisk.org/jira/browse/ASTERISK-27480>] - Security: Authenticated SUBSCRIBE without Contact crashes asterisk (Reported by Ross Beer) - [ASTERISK-27299 <https://issues.asterisk.org/jira/browse/ASTERISK-27299>] - Asterisk Hangs with Bad file descriptor on read() (Reported by Abhay Gupta) - [ASTERISK-25079 <https://issues.asterisk.org/jira/browse/ASTERISK-25079>] - AMI bridge of channels results in MOH not destroyed and robotic audio on one channel (Reported by Zane Conkle) - [ASTERISK-27490 <https://issues.asterisk.org/jira/browse/ASTERISK-27490>] - chan_console: 'set active' fails to work (Reported by Tzafrir Cohen) - [ASTERISK-24756 <https://issues.asterisk.org/jira/browse/ASTERISK-24756>] - ConfBridge sound_muted does not work from CLI or AMI (Reported by Thomas Frederiksen) - [ASTERISK-25649 <https://issues.asterisk.org/jira/browse/ASTERISK-25649>] - Transfer application does not work with Local channels - documentation misleading (Reported by Ivan Ullmann) - [ASTERISK-25869 <https://issues.asterisk.org/jira/browse/ASTERISK-25869>] - chan_sip: "rejected because extension not found" should be logged as a security event (Reported by Brian J. Murrell) - [ASTERISK-27440 <https://issues.asterisk.org/jira/browse/ASTERISK-27440>] - Strictrtp has issues to qualify video rtp streams (Reported by Wim De Vlaminck) - [ASTERISK-24329 <https://issues.asterisk.org/jira/browse/ASTERISK-24329>] - Music On Hold announcement cuts intro of music the first time it is played (Reported by Thomas Frederiksen) - [ASTERISK-19657 <https://issues.asterisk.org/jira/browse/ASTERISK-19657>] - Coverity Report: Fix issues for error type CHAR_IO (Reported by Matt Jordan) - [ASTERISK-27175 <https://issues.asterisk.org/jira/browse/ASTERISK-27175>] - iax.conf demo peer is invalid (Reported by Tzafrir Cohen) - [ASTERISK-27430 <https://issues.asterisk.org/jira/browse/ASTERISK-27430>] - README refers to security documents that do not exist. (Reported by Corey Farrell) - [ASTERISK-20281 <https://issues.asterisk.org/jira/browse/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 <https://issues.asterisk.org/jira/browse/ASTERISK-27382>] - crash after an invalid rtcp packet from GT48 FXS gateway (Reported by Tzafrir Cohen) - [ASTERISK-27429 <https://issues.asterisk.org/jira/browse/ASTERISK-27429>] - res_rtp_asterisk: Multiple reports in an RTCP packet will write past where it should (Reported by Vitezslav Novy) - [ASTERISK-27408 <https://issues.asterisk.org/jira/browse/ASTERISK-27408>] - Identify causes and fix pjsip/resolver/srv/failover/in_dialog/transport_tcp (Reported by Corey Farrell) - [ASTERISK-18411 <https://issues.asterisk.org/jira/browse/ASTERISK-18411>] - Queue members with hints for state_interface get stuck in "In Use" state. (Reported by Steven T. Wheeler) - [ASTERISK-26131 <https://issues.asterisk.org/jira/browse/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 <https://issues.asterisk.org/jira/browse/ASTERISK-27475>] - codec_opus requires libcurl (Reported by Samuel For) - [ASTERISK-27467 <https://issues.asterisk.org/jira/browse/ASTERISK-27467>] - pjsip_options: qualify_frequency sometimes not applied on reload (Reported by John Bigelow) - [ASTERISK-27465 <https://issues.asterisk.org/jira/browse/ASTERISK-27465>] - CLI Completion Not Working (Reported by Ross Beer) - [ASTERISK-27460 <https://issues.asterisk.org/jira/browse/ASTERISK-27460>] - CDR: Deadlock using AMI Originate with Variable CDR(amaflags)=... (Reported by Richard Mudgett) - [ASTERISK-27453 <https://issues.asterisk.org/jira/browse/ASTERISK-27453>] - RTP: Blind transfer direct media scenario results in one way audio. (Reported by Richard Mudgett) - [ASTERISK-20643 <https://issues.asterisk.org/jira/browse/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 <https://issues.asterisk.org/jira/browse/ASTERISK-26980>] - pjsip: Clean up WebRTC disables (Reported by abelbeck) - [ASTERISK-27452 <https://issues.asterisk.org/jira/browse/ASTERISK-27452>] - Security: chan_skinny: Memory exhaustion if flooded with unauthenticated requests (Reported by George Joseph) - [ASTERISK-27454 <https://issues.asterisk.org/jira/browse/ASTERISK-27454>] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported by Joshua Colp) - [ASTERISK-23735 <https://issues.asterisk.org/jira/browse/ASTERISK-23735>] - Transcoding makes bad choice in high-rate translations (Reported by Richard Kenner) - [ASTERISK-27445 <https://issues.asterisk.org/jira/browse/ASTERISK-27445>] - ARI: Updating a bridge gives wrong error message. (Reported by Frank Durden) - [ASTERISK-24662 <https://issues.asterisk.org/jira/browse/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 <https://issues.asterisk.org/jira/browse/ASTERISK-27353>] - H323 audio starts with a delay of 2 seconds. (Reported by Marco Giordani) - [ASTERISK-27442 <https://issues.asterisk.org/jira/browse/ASTERISK-27442>] - pjsip: 183 without To tag does not negotiate media (Reported by Kevin Harwell) - [ASTERISK-27437 <https://issues.asterisk.org/jira/browse/ASTERISK-27437>] - [patch] ICE: server-reflexive candidates (srflx) with Dual-Stack. (Reported by Alexander Traud) - [ASTERISK-27434 <https://issues.asterisk.org/jira/browse/ASTERISK-27434>] - [patch] chan_sip/ICE: Square brackets around IPv6 addresses. (Reported by Alexander Traud) - [ASTERISK-27435 <https://issues.asterisk.org/jira/browse/ASTERISK-27435>] - [patch] configure: pjsip_evsub_set_uas_timeout not found. (Reported by Alexander Traud) - [ASTERISK-27431 <https://issues.asterisk.org/jira/browse/ASTERISK-27431>] - Asterisk fails to build when openssl headers are not installed. (Reported by Corey Farrell) - [ASTERISK-27332 <https://issues.asterisk.org/jira/browse/ASTERISK-27332>] - Asterisk fails to configure on MacOS Sierra (Reported by Ivan Larionov) - [ASTERISK-27421 <https://issues.asterisk.org/jira/browse/ASTERISK-27421>] - RTP source learning not working with devices that have some clock issues (Reported by nappsoft) - [ASTERISK-27361 <https://issues.asterisk.org/jira/browse/ASTERISK-27361>] - Attended transfer crashes in Asterisk 13.17.2 (Reported by Alessandro Pimenta) - [ASTERISK-27238 <https://issues.asterisk.org/jira/browse/ASTERISK-27238>] - Bridging: Crash freeing a frame that's already been freed (Reported by Richard Kenner) - [ASTERISK-27412 <https://issues.asterisk.org/jira/browse/ASTERISK-27412>] - core: Audiohook freeing interpolated frame when it shouldn't. (Reported by Mikhail) - [ASTERISK-27423 <https://issues.asterisk.org/jira/browse/ASTERISK-27423>] - app_record: We set the RECORD_STATUS channel variable before closing the file (Reported by George Joseph) - [ASTERISK-26758 <https://issues.asterisk.org/jira/browse/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 <https://issues.asterisk.org/jira/browse/ASTERISK-27363>] - res_http_websocket: Wrong LocalAddress (it is equal to RemoteAddress) (Reported by Vasilii Rogin) - [ASTERISK-27415 <https://issues.asterisk.org/jira/browse/ASTERISK-27415>] - asterisk.conf: Setting astctl without setting astrundir is ineffective. (Reported by Corey Farrell) - [ASTERISK-27411 <https://issues.asterisk.org/jira/browse/ASTERISK-27411>] - pjsip: TCP connections may not be destroyed (Reported by Joshua Colp) - [ASTERISK-27345 <https://issues.asterisk.org/jira/browse/ASTERISK-27345>] - res_pjsip_session: RTP instances leak on 488 responses. (Reported by Corey Farrell) - [ASTERISK-27337 <https://issues.asterisk.org/jira/browse/ASTERISK-27337>] - chan_sip: Security vulnerability with client code header (revisited) (Reported by Richard Mudgett) - [ASTERISK-27319 <https://issues.asterisk.org/jira/browse/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 <https://issues.asterisk.org/jira/browse/ASTERISK-27391>] - Regression: Deadlock between AOR named lock and pjproject grp lock (Reported by shaurya jain) - [ASTERISK-27393 <https://issues.asterisk.org/jira/browse/ASTERISK-27393>] - res_pjsip: Crash occurs when an empty contact read from astdb or database (Reported by Aaron An) - [ASTERISK-27290 <https://issues.asterisk.org/jira/browse/ASTERISK-27290>] - res_pjsip: PIDF contact field has malformed/invalid XML (Reported by basildane) - [ASTERISK-27032 <https://issues.asterisk.org/jira/browse/ASTERISK-27032>] - res_pjsip: TLS options do not handle empty values (Reported by seanchann.zhou) - [ASTERISK-27394 <https://issues.asterisk.org/jira/browse/ASTERISK-27394>] - [patch] tcptls: Print notice when TLS is enabled but not configured. (Reported by Alexander Traud) - [ASTERISK-26426 <https://issues.asterisk.org/jira/browse/ASTERISK-26426>] - format_ogg_opus: remove from source (Reported by Kevin Harwell) - [ASTERISK-27378 <https://issues.asterisk.org/jira/browse/ASTERISK-27378>] - Modules: Fix issues with CLI completion. (Reported by Corey Farrell) - [ASTERISK-27387 <https://issues.asterisk.org/jira/browse/ASTERISK-27387>] - Regression: pjsip 13.18.0 - from_user - "+" character isn't allowed any more (Reported by Michael Maier) - [ASTERISK-27390 <https://issues.asterisk.org/jira/browse/ASTERISK-27390>] - Audit menuselect module dependencies (Reported by Corey Farrell) - [ASTERISK-27389 <https://issues.asterisk.org/jira/browse/ASTERISK-27389>] - Optional API modules should not allow unload. (Reported by Corey Farrell) - [ASTERISK-27369 <https://issues.asterisk.org/jira/browse/ASTERISK-27369>] - Bridge() dialplan application fails without setting BRIDGERESULT channel variable (Reported by James Terhune) - [ASTERISK-27377 <https://issues.asterisk.org/jira/browse/ASTERISK-27377>] - Typo in CHANNEL(dtmf_features) usage documentation (Reported by Igor Goncharovsky) - [ASTERISK-27181 <https://issues.asterisk.org/jira/browse/ASTERISK-27181>] - GCC 7 warning: app_voicemail.c: In function 'imap_delete_old_greeting' (Reported by Anthony Messina) - [ASTERISK-27194 <https://issues.asterisk.org/jira/browse/ASTERISK-27194>] - jitterbuffer: Does not handle case where translator returns null frame. (Reported by Joshua Elson) - [ASTERISK-26639 <https://issues.asterisk.org/jira/browse/ASTERISK-26639>] - core: Disabling xmldoc support does not work. Also results in abort during Asterisk startup. (Reported by Mr Dini) - [ASTERISK-27372 <https://issues.asterisk.org/jira/browse/ASTERISK-27372>] - ARI: Node ARI client broken in latest versions of 13 and 14 (Reported by Benjamin Keith Ford) - [ASTERISK-18140 <https://issues.asterisk.org/jira/browse/ASTERISK-18140>] - Expires handling in SUBSCRIBE confuses the absence of the Expires header field with an unsubscribe action. (Reported by Jonathan Cloots) - [ASTERISK-25960 <https://issues.asterisk.org/jira/browse/ASTERISK-25960>] - The config_hook unit test causes Asterisk to crash if run a second time (Reported by George Joseph) - [ASTERISK-27198 <https://issues.asterisk.org/jira/browse/ASTERISK-27198>] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6 set to yes (Reported by Martin Cisárik) - [ASTERISK-27346 <https://issues.asterisk.org/jira/browse/ASTERISK-27346>] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded (Reported by Ronald Raikes) - [ASTERISK-27365 <https://issues.asterisk.org/jira/browse/ASTERISK-27365>] - [patch] chan_sip: Crypto attribute not last but first on SDP media level. (Reported by Alexander Traud) - [ASTERISK-24483 <https://issues.asterisk.org/jira/browse/ASTERISK-24483>] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on un/re-load: mod.id == -1 (Reported by Tzafrir Cohen) - [ASTERISK-23462 <https://issues.asterisk.org/jira/browse/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-27328 <https://issues.asterisk.org/jira/browse/ASTERISK-27328>] - Missing openssl dependencies in res_rtp_asterisk and tcptls (Reported by Tzafrir Cohen) - [ASTERISK-27341 <https://issues.asterisk.org/jira/browse/ASTERISK-27341>] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains local address. (Reported by Alexander Traud) - [ASTERISK-27343 <https://issues.asterisk.org/jira/browse/ASTERISK-27343>] - Fails to build in FreeBSD due to sys/sysmacros.h not existing there (Reported by Guido Falsi) - [ASTERISK-27340 <https://issues.asterisk.org/jira/browse/ASTERISK-27340>] - backtrace.c: Crash due to double-free. (Reported by Corey Farrell) - [ASTERISK-27339 <https://issues.asterisk.org/jira/browse/ASTERISK-27339>] - [patch] Crash on ast_ssl_teardown when stopping. (Reported by Alexander Traud) - [ASTERISK-27333 <https://issues.asterisk.org/jira/browse/ASTERISK-27333>] - sip_to_pjsip not correctly handling disallow=all directive (Reported by Torrey Searle) *Improvements made in this release:* ----------------------------------- - [ASTERISK-24297 <https://issues.asterisk.org/jira/browse/ASTERISK-24297>] - cdr.c: Minor code optimizations. (Reported by Richard Mudgett) - [ASTERISK-27449 <https://issues.asterisk.org/jira/browse/ASTERISK-27449>] - [PATCH] When failing to acquire target during attended transfer, display wanted extension (Reported by Niklas Larsson) - [ASTERISK-27456 <https://issues.asterisk.org/jira/browse/ASTERISK-27456>] - app_voicemail: Add new object for VoicemailUserEntry (Reported by sungtae kim) - [ASTERISK-27380 <https://issues.asterisk.org/jira/browse/ASTERISK-27380>] - ast_coredumper: allow pointing out the asterisk binary explicitly (Reported by Tzafrir Cohen) - [ASTERISK-23556 <https://issues.asterisk.org/jira/browse/ASTERISK-23556>] - Compilation warning for invert.c (array subscript is above array bounds) (Reported by Marcello Ceschia) - [ASTERISK-27355 <https://issues.asterisk.org/jira/browse/ASTERISK-27355>] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard Mudgett) - [ASTERISK-27335 <https://issues.asterisk.org/jira/browse/ASTERISK-27335>] - CDR performance needs improvement. (Reported by Richard Mudgett) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.19.0 *Thank you for your continued support of Asterisk!* -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
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