The Asterisk Development Team would like to announce the first release candidate of Asterisk 18.16.0. This release candidate is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 18.16.0-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: Security bugs fixed in this release: ----------------------------------- * ASTERISK-30338 - pjproject: Backport security fixes from 2.13 (Reported by Benjamin Keith Ford) * ASTERISK-30176 - manager: GetConfig can read files outside of Asterisk (Reported by shawty) * ASTERISK-30103 - chan_ooh323 Vulnerability in calling/called party IE (Reported by Michael Bradeen) Improvements made in this release: ----------------------------------- * ASTERISK-30328 - Typo in from_domain description on res_pjsip configuration documentation (Reported by Marcel Wagner) * ASTERISK-30316 - res_pjsip: Documentation should point out default if contact_user is not being set for outbound registrations (Reported by Marcel Wagner) * ASTERISK-30289 - xmldoc: Allow XML docs to be reloaded (Reported by N A) * ASTERISK-30327 - rtp_engine.h: Remove obsolete example usage (Reported by N A) * ASTERISK-30286 - app_mixmonitor: Add option to use real Caller ID for Caller ID (Reported by N A) * ASTERISK-30308 - pbx_builtins: Allow Answer to return immediately (Reported by N A) * ASTERISK-30295 - test_json: Remove duplicated static function (Reported by N A) * ASTERISK-30290 - file.c: Don't emit warnings on winks. (Reported by N A) * ASTERISK-30241 - res_pjsip_gelocation: Downgrade some NOTICE scope trace debugs to DEBUG level (Reported by N A) * ASTERISK-30223 - features: add no-answer option to Bridge application (Reported by N A) * ASTERISK-30158 - PJSIP: Add new 100rel option "peer_supported" (Reported by Maximilian Fridrich) Bugs fixed in this release: ----------------------------------- * ASTERISK-30349 - app_if: Format truncation error (Reported by George Joseph) * ASTERISK-30265 - res_pjsip_session: Fix missing PLAR support on INVITEs (Reported by N A) * ASTERISK-30283 - app_voicemail: Fix msg_create_from_file not sending email to user (Reported by N A) * ASTERISK-29793 - adsi: CAS is malformed (Reported by N A) * ASTERISK-30344 - ari: Memory leak in create when specifying JSON (Reported by Saken) * ASTERISK-30311 - func_presencestate: Fix invalid memory access. (Reported by N A) * ASTERISK-30336 - sig_analog: Fix no timeout duration (Reported by N A) * ASTERISK-30244 - res_pjsip_pubsub: Occasional crash when TCP/TLS connection terminated and subscription persistence is removed (Reported by nappsoft) * ASTERISK-30184 - res_pjsip_session: re-INVITE after answering results in wrong stream direction of first call leg (Reported by Maximilian Fridrich) * ASTERISK-29998 - sla: deadlock when calling SLAStation application (Reported by N A) * ASTERISK-30321 - Build: Embedded blobs have executable stacks (Reported by George Joseph) * ASTERISK-30293 - Memory leak in JSON_DECODE (Reported by David Uczen) * ASTERISK-30314 - res_agi: RECORD FILE doesn't respect "transmit_silence" asterisk.conf option (Reported by Joshua C. Colp) * ASTERISK-30285 - manager.c: Remove outdated documentation (Reported by N A) * ASTERISK-30076 - app_stack: Incorrect exit location in predial handlers logged (Reported by N A) * ASTERISK-30282 - CI: Coredump output isn't saved when running unittests (Reported by George Joseph) * ASTERISK-30281 - chan_rtp: Local address being used before being set (Reported by George Joseph) * ASTERISK-28689 - res_pjsip: Crash when locking group lock when sending stateful response (Reported by Jesse Ross) * ASTERISK-30217 - Registration do not allow multiple proxies (Reported by Igor Goncharovsky) * ASTERISK-30278 - tcptls: Abort occurs if SSL error is logged if MALLOC_DEBUG is enabled (Reported by N A) * ASTERISK-30273 - test_mwi: compilation fails on 32-bit Debian (Reported by N A) * ASTERISK-30193 - chan_pjsip should return all codecs on a re-INVITE without SDP (Reported by Henning Westerholt) * ASTERISK-30258 - Dialing API: Cancel a running async thread, does not always cancel all calls (Reported by Frederic LE FOLL) * ASTERISK-30274 - chan_dahdi: Unavailable channels are BUSY (Reported by N A) * ASTERISK-30248 - ast_get_digit_str adds bogus initial delimiter if first character not to be spoken (Reported by David Woolley) * ASTERISK-30264 - res_pjsip: Subscription handlers do not get cleanly unregistered, causing crash (Reported by N A) * ASTERISK-30213 - Make crypto_load() reentrant and handle symlinks correctly (Reported by Philip Prindeville) * ASTERISK-30256 - chan_dahdi: Fix format truncation warnings (Reported by N A) * ASTERISK-30239 - Prometheus plugin crashes Asterisk when using local channel (Reported by Joeran Vinzens) * ASTERISK-30237 - res_prometheus: Crash when scraping bridges (Reported by Igor Yeroshev) * ASTERISK-30245 - db: ListItems is incorrect (Reported by N A) * ASTERISK-30243 - func_logic: IF function complains if both branches are empty (Reported by N A) * ASTERISK-30232 - Initialize stack-based ast_test_capture structures correctly (Reported by Philip Prindeville) * ASTERISK-30220 - func_scramble: Fix segfault due to null pointer deref (Reported by N A) * ASTERISK-30235 - res_crypto and tests: Memory issues and and uninitialized variable error (Reported by George Joseph) * ASTERISK-30234 - res_geolocation: ...may be used uninitialized error in geoloc_config.c (Reported by George Joseph) * ASTERISK-30226 - REGRESSION: res_crypto complains about the stir_shaken directory in /var/lib/asterisk/keys (Reported by George Joseph) New Features made in this release: ----------------------------------- * ASTERISK-21502 - New SIP Channel Driver - add Advice of Charge support (Reported by Matt Jordan) * ASTERISK-30150 - res_pjsip_session: Add support for custom parameters (Reported by N A) * ASTERISK-30322 - res_hep: Add capture agent name support (Reported by N A) * ASTERISK-29497 - Add conditional branch applications (Reported by N A) * ASTERISK-30305 - chan_dahdi: Allow FXO channels to start immediately (Reported by N A) * ASTERISK-30284 - app_mixmonitor: Add option to delete recording file when done (Reported by N A) * ASTERISK-30146 - res_pjsip_logger: Add method-based log filtering (Reported by N A) * ASTERISK-30263 - res_pjsip_notify: Allow using pjsip_notify.conf from AMI (Reported by N A) * ASTERISK-30254 - res_tonedetect: Add audible ringback detection to TONE_DETECT (Reported by N A) * ASTERISK-30091 - cdr: Allow CDRs to ignore call state changes (Reported by N A) * ASTERISK-30032 - Support of mediasec SIP headers and SDP attributes (Reported by Maximilian Fridrich) * ASTERISK-30216 - app_bridgewait: Add option for BridgeWait to not answer (Reported by N A) * ASTERISK-30179 - app_amd: Allow audio to be played while AMD is running (Reported by N A) * ASTERISK-29432 - New function to allow access to any channel (Reported by N A) * ASTERISK-30222 - func_strings: Add trim functions (Reported by N A) For a full list of changes in this release candidate, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0-rc1 Thank you for your continued support of Asterisk!
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