The Asterisk Development Team would like to announce the release of Asterisk 20.1.0.
This release is available for immediate download at
https://downloads.asterisk.
The release of Asterisk 20.1.0 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:
Security bugs fixed in this release:
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pjproject: Backport security fixes from 2.13 (Reported by Benjamin Keith Ford) |
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manager: GetConfig can read files outside of Asterisk (Reported by shawty) |
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chan_ooh323 Vulnerability in calling/called party IE (Reported by Michael Bradeen) |
Improvements made in this release:
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Typo in from_domain description on res_pjsip configuration documentation (Reported by Marcel Wagner) |
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res_pjsip: Documentation should point out default if contact_user is not being set for outbound registrations (Reported by Marcel Wagner) |
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xmldoc: Allow XML docs to be reloaded (Reported by N A) |
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rtp_engine.h: Remove obsolete example usage (Reported by N A) |
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app_mixmonitor: Add option to use real Caller ID for Caller ID (Reported by N A) |
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pbx_builtins: Allow Answer to return immediately (Reported by N A) |
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test_json: Remove duplicated static function (Reported by N A) |
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file.c: Don’t emit warnings on winks. (Reported by N A) |
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res_pjsip_gelocation: Downgrade some NOTICE scope trace debugs to DEBUG level (Reported by N A) |
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features: add no-answer option to Bridge application (Reported by N A) |
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PJSIP: Add new 100rel option “peer_supported” (Reported by Maximilian Fridrich) |
Bugs fixed in this release:
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res_pjsip: Websockets from same IP shut down when they shouldn’t be (Reported by Joshua C. Colp) |
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app_if: Format truncation error (Reported by George Joseph) |
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ari: Memory leak in create when specifying JSON (Reported by Saken) |
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app_voicemail: Fix msg_create_from_file not sending email to user (Reported by N A) |
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res_pjsip_session: Fix missing PLAR support on INVITEs (Reported by N A) |
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adsi: CAS is malformed (Reported by N A) |
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func_presencestate: Fix invalid memory access. (Reported by N A) |
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sig_analog: Fix no timeout duration (Reported by N A) |
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res_pjsip_pubsub: Occasional crash when TCP/TLS connection terminated and subscription persistence is removed (Reported by nappsoft) |
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res_pjsip_session: re-INVITE after answering results in wrong stream direction of first call leg (Reported by Maximilian Fridrich) |
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sla: deadlock when calling SLAStation application (Reported by N A) |
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Build: Embedded blobs have executable stacks (Reported by George Joseph) |
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Memory leak in JSON_DECODE (Reported by David Uczen) |
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res_agi: RECORD FILE doesn’t respect “transmit_silence” asterisk.conf option (Reported by Joshua C. Colp) |
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manager.c: Remove outdated documentation (Reported by N A) |
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CI: Coredump output isn’t saved when running unittests (Reported by George Joseph) |
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app_stack: Incorrect exit location in predial handlers logged (Reported by N A) |
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chan_rtp: Local address being used before being set (Reported by George Joseph) |
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res_pjsip: Crash when locking group lock when sending stateful response (Reported by Jesse Ross) |
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tcptls: Abort occurs if SSL error is logged if MALLOC_DEBUG is enabled (Reported by N A) |
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Registration do not allow multiple proxies (Reported by Igor Goncharovsky) |
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test_mwi: compilation fails on 32-bit Debian (Reported by N A) |
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chan_pjsip should return all codecs on a re-INVITE without SDP (Reported by Henning Westerholt) |
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Dialing API: Cancel a running async thread, does not always cancel all calls (Reported by Frederic LE FOLL) |
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chan_dahdi: Unavailable channels are BUSY (Reported by N A) |
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res_pjsip: Subscription handlers do not get cleanly unregistered, causing crash (Reported by N A) |
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ast_get_digit_str adds bogus initial delimiter if first character not to be spoken (Reported by David Woolley) |
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Make crypto_load() reentrant and handle symlinks correctly (Reported by Philip Prindeville) |
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chan_dahdi: Fix format truncation warnings (Reported by N A) |
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Prometheus plugin crashes Asterisk when using local channel (Reported by Joeran Vinzens) |
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res_prometheus: Crash when scraping bridges (Reported by Igor Yeroshev) |
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db: ListItems is incorrect (Reported by N A) |
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func_logic: IF function complains if both branches are empty (Reported by N A) |
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Initialize stack-based ast_test_capture structures correctly (Reported by Philip Prindeville) |
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func_scramble: Fix segfault due to null pointer deref (Reported by N A) |
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res_crypto and tests: Memory issues and and uninitialized variable error (Reported by George Joseph) |
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res_geolocation: …may be used uninitialized error in geoloc_config.c (Reported by George Joseph) |
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REGRESSION: res_crypto complains about the stir_shaken directory in /var/lib/asterisk/keys (Reported by George Joseph) |
New Features made in this release:
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New SIP Channel Driver – add Advice of Charge support (Reported by Matt Jordan) |
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res_hep: Add capture agent name support (Reported by N A) |
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Add conditional branch applications (Reported by N A) |
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res_pjsip_session: Add support for custom parameters (Reported by N A) |
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chan_dahdi: Allow FXO channels to start immediately (Reported by N A) |
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app_mixmonitor: Add option to delete recording file when done (Reported by N A) |
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res_pjsip_notify: Allow using pjsip_notify.conf from AMI (Reported by N A) |
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res_pjsip_logger: Add method-based log filtering (Reported by N A) |
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cdr: Allow CDRs to ignore call state changes (Reported by N A) |
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res_tonedetect: Add audible ringback detection to TONE_DETECT (Reported by N A) |
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Support of mediasec SIP headers and SDP attributes (Reported by Maximilian Fridrich) |
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app_bridgewait: Add option for BridgeWait to not answer (Reported by N A) |
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app_amd: Allow audio to be played while AMD is running (Reported by N A) |
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New function to allow access to any channel (Reported by N A) |
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func_strings: Add trim functions (Reported by N A) |
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.
Thank you for your continued support of Asterisk!