The Asterisk Development Team would like to announce security release Asterisk 21.0.1. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/21.0.1 and https://downloads.asterisk.org/pub/telephony/asterisk The following security advisories were resolved in this release: - [Path traversal via AMI GetConfig allows access to outside files](https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f) - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq) - [PJSIP logging allows attacker to inject fake Asterisk log entries ](https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7) - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh) Change Log for Release asterisk-21.0.1 ======================================== Links: ---------------------------------------- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md) - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1) - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz) - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) Summary: ---------------------------------------- - res_pjsip_header_funcs: Duplicate new header value, don't copy. - res_pjsip: disable raw bad packet logging - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list - manager.c: Prevent path traversal with GetConfig. User Notes: ---------------------------------------- - ### http.c: Minor simplification to HTTP status output. For bound addresses, the HTTP status page now combines the bound address and bound port in a single line. Additionally, the SSL bind address has been renamed to TLS. Upgrade Notes: ---------------------------------------- - ### chan_sip: Remove deprecated module. This module was deprecated in Asterisk 17 and is now being removed in accordance with the Asterisk Module Deprecation policy. - ### res_monitor: Remove deprecated module. This module was deprecated in Asterisk 16 and is now being removed in accordance with the Asterisk Module Deprecation policy. This also removes the 'w' and 'W' options for app_queue. MixMonitor should be default and only option for all settings that previously used either Monitor or MixMonitor. - ### app_osplookup: Remove deprecated module. This module was deprecated in Asterisk 19 and is now being removed in accordance with the Asterisk Module Deprecation policy. - ### app_cdr: Remove deprecated application and option. The previously deprecated NoCDR application has been removed. Additionally, the previously deprecated 'e' option to the ResetCDR application has been removed. - ### chan_skinny: Remove deprecated module. This module was deprecated in Asterisk 19 and is now being removed in accordance with the Asterisk Module Deprecation policy. - ### chan_mgcp: Remove deprecated module. This module was deprecated in Asterisk 19 and is now being removed in accordance with the Asterisk Module Deprecation policy. - ### translate.c: Prefer better codecs upon translate ties. When setting up translation between two codecs the quality was not taken into account, resulting in suboptimal translation. The quality is now taken into account, which can reduce the number of translation steps required, and improve the resulting quality. - ### app_macro: Remove deprecated module. This module was deprecated in Asterisk 16 and is now being removed in accordance with the Asterisk Module Deprecation policy. For most modules that interacted with app_macro, this change is limited to no longer looking for the current context from the macrocontext when set. The following modules have additional impacts: app_dial - no longer supports M^ connected/redirecting macro app_minivm - samples written using macro will no longer work. The sample needs to be re-written app_queue - can no longer call a macro on the called party's channel. Use gosub which is currently supported ccss - no callback macro, gosub only app_voicemail - no macro support channel - remove macrocontext and priority, no connected line or redirection macro options options - stdexten is deprecated to gosub as the default and only options pbx - removed macrolock pbx_dundi - no longer look for macro snmp - removed macro context, exten, and priority - ### chan_alsa: Remove deprecated module. This module was deprecated in Asterisk 19 and is now being removed in accordance with the Asterisk Module Deprecation policy. - ### pbx_builtins: Remove deprecated and defunct functionality. The previously deprecated ImportVar and SetAMAFlags applications have now been removed. Closed Issues: ---------------------------------------- None -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-announce mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-announce
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