The Asterisk Development Team would like to announce the release of Asterisk 20.4.0.
The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.4.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You! Change Log for Release 20.4.0 ======================================== Links: ---------------------------------------- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md) - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz) - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) Summary: ---------------------------------------- - app.h: Move declaration of ast_getdata_result before its first use - doc: Remove obsolete CHANGES-staging and UPGRADE-staging - .github: Updates for AsteriskReleaser - app_voicemail: fix imap compilation errors - res_musiconhold: avoid moh state access on unlocked chan - utils: add lock timestamps for DEBUG_THREADS - .github: Back out triggering PROpenedOrUpdated by label - .github: Move publish docs to new file CreateDocs.yml - rest-api: Updates for new documentation site - .github: Remove result check from PROpenUpdateGateTests - .github: Fix use of 'contains' - .github: Add recheck label test to additional jobs - .github: Fix recheck label typos - .github: Fix recheck label manipulation - .github: Allow PR submit checks to be re-run by label - app_voicemail_imap: Fix message count when IMAP server is unavailable - res_pjsip_rfc3326: Prefer Q.850 cause code over SIP. - res_pjsip_session: Added new function calls to avoid ABI issues. - app_queue: Add force_longest_waiting_caller option. - pjsip_transport_events.c: Use %zu printf specifier for size_t. - res_crypto.c: Gracefully handle potential key filename truncation. - configure: Remove obsolete and deprecated constructs. - res_fax_spandsp.c: Clean up a spaces/tabs issue - ast-db-manage: Synchronize revisions between comments and code. - test_statis_endpoints: Fix channel_messages test again - res_crypto.c: Avoid using the non-portable ALLPERMS macro. - tcptls: when disabling a server port, we should set the accept_fd to -1. - AMI: Add parking position parameter to Park action - test_stasis_endpoints.c: Make channel_messages more stable - build: Fix a few gcc 13 issues - .github: Rework for merge approval - ast-db-manage: Fix alembic branching error caused by #122. - app_followme: fix issue with enable_callee_prompt=no (#88) - sounds: Update download URL to use HTTPS. - configure: Makefile downloader enable follow redirects. - res_musiconhold: Add option to loop last file. - chan_dahdi: Fix Caller ID presentation for FXO ports. - AMI: Add CoreShowChannelMap action. - sig_analog: Add fuller Caller ID support. - res_stasis.c: Add new type 'sdp_label' for bridge creation. - app_queue: Preserve reason for realtime queues - .github: Fix issues with cherry-pick-reminder - indications: logging changes - .github Ignore error when adding reviewrs to PR - .github: Update field descriptions for AsteriskReleaser - callerid: Allow specifying timezone for date/time. - logrotate: Fix duplicate log entries. - chan_pjsip: Allow topology/session refreshes in early media state - chan_dahdi: Fix broken hidecallerid setting. - .github: Change title of AsteriskReleaser job - asterisk.c: Fix option warning for remote console. - .github: Don't add cherry-pick reminder if it's already present - .github: Fix quoting in PROpenedOrUpdated - .github: Add cherry-pick reminder to new PRs - configure: fix test code to match gethostbyname_r prototype. - res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77) - res_sorcery_memory_cache.c: Fix memory leak - xml.c: Process XML Inclusions recursively. - .github: Tweak improvement issue type language. - .github: Tweak new feature language, and move feature requests elsewhere. - .github: Fix staleness check to only run on certain labels. User Notes: ---------------------------------------- - ### AMI: Add parking position parameter to Park action New ParkingSpace parameter has been added to AMI action Park. - ### res_musiconhold: Add option to loop last file. The loop_last option in musiconhold.conf now allows the last file in the directory to be looped once reached. - ### AMI: Add CoreShowChannelMap action. New AMI action CoreShowChannelMap has been added. - ### sig_analog: Add fuller Caller ID support. Additional Caller ID properties are now supported on incoming calls to FXS stations, namely the redirecting reason and call qualifier. - ### res_stasis.c: Add new type 'sdp_label' for bridge creation. When creating a bridge using the ARI the 'type' argument now accepts a new value 'sdp_label' which will configure the bridge to add labels for each stream in the SDP with the corresponding channel id. - ### app_queue: Preserve reason for realtime queues Make paused reason in realtime queues persist an Asterisk restart. This was fixed for non-realtime queues in ASTERISK_25732. Upgrade Notes: ---------------------------------------- - ### app_queue: Preserve reason for realtime queues Add a new column to the queue_member table: reason_paused VARCHAR(80) so the reason can be preserved. Closed Issues: ---------------------------------------- - #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is insufficient/ineffective - #55: [bug]: res_sorcery_memory_cache: Memory leak when calling sorcery_memory_cache_open - #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection - #65: [bug]: heap overflow by default at startup - #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for realtime queues - #73: [new-feature]: pjsip: Allow topology/session refreshes in early media state - #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout - #89: [improvement]: indications: logging changes - #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP labels - #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls - #96: [bug]: make install-logrotate causes logrotate to fail on service restart - #98: [new-feature]: callerid: Allow timezone to be specified at runtime - #100: [bug]: sig_analog: hidecallerid setting is broken - #102: [bug]: Strange warning - 'T' option is not compatible with remote console mode and has no effect. - #104: [improvement]: Add AMI action to get a list of connected channels - #108: [new-feature]: fair handling of calls in multi-queue scenarios - #110: [improvement]: utils - add lock timing information with DEBUG_THREADS - #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating - #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID - #122: [new-feature]: res_musiconhold: Add looplast option - #133: [bug]: unlock channel after moh state access - #136: [bug]: Makefile downloader does not follow redirects. - #145: [bug]: ABI issue with pjproject and pjsip_inv_session - #155: [bug]: GCC 13 is catching a few new trivial issues - #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable - #174: [bug]: app_voicemail imap compile errors - #200: [bug]: Regression: In app.h an enum is used before its declaration. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? 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