asterisk 22.1.0

Asterisk is an open source framework for building communications applications. Asterisk turns an ordinary computer into a communications server. Asterisk powers IP PBX systems, VoIP gateways, conference servers and other custom solutions. It is used by small businesses, large businesses, call centers, carriers and government agencies, worldwide. Asterisk is free and open source.

Tags communication conferencing telephony sip pbx c python
License GNU GPL
State stable

Recent Releases

22.1.030 Nov 2024 18:25 minor feature:
18.25.027 Nov 2024 11:45 minor feature:
21.4.222 Sep 2024 04:05 minor feature:
21.4.119 Aug 2024 21:45 minor feature:
21.4.001 Aug 2024 01:05 minor feature:
21.3.121 May 2024 19:45 minor feature:
21.3.017 May 2024 12:45 minor feature:
21.2.019 Mar 2024 19:25 minor feature: Res_pjsip_stir_shaken.c: Add checks for missing parameters. App_dial: Add dial time for progress/ringing. App_voicemail: Properly reinitialize config after unit tests. App_queue.c : "queue add member" usage string. App_voicemail: Allow preventing mark messages as urgent. Res_pjsip: Use consistent type for boolean columns. Attestation_config.c: Use ast_free instead of ast_std_free. Makefile: Add stir_shaken/cache to directories created on install. Stir/Shaken Refactor. Translate.c: implement new direct comp table mode. README.md: Removed outdated link. Strings.h: Ensure ast_str_buffer( ) returns a 0 terminated string. Res_rtp_asterisk.c: Correct coefficient in MOS calculation. Dsp.c: and improve potentially inaccurate log message. Pjsip show channelstats: Prevent possible segfault when faxing. Reduce startup/shutdown verbose logging. Configure: Rerun bootstrap on modern platform. Upgrade bundled pjproject to 2.14. Res_pjsip_outbound_registration.c: Add User-Agent header override. App_speech_utils.c: Allow partial speech results. Utils: Make behavior of ast_strsep match strsep. App_chanspy: Add 'D' option for dual-channel audio. App_if: next priority calculation. Res_pjsip_t38.c: Permit IPv6 SDP connection addresses. BuildSystem: Bump autotools versions on OpenBSD. Main/utils: Simplify the FreeBSD ast_get_tid() handling. Res_pjsip_session.c: Correctly format SDP connection addresses. Rtp_engine.c: Correct sample rate typo for L16/44100. Manager.c: erroneous reloads in UpdateConfig. Res_calendar_icalendar: Print iCalendar error on parsing failure. App_confbridge: Don't emit warnings on valid configurations. App_voicemail_odbc: remove macrocontext from voicemail_messages table. Chan_dahdi: Allow MWI to be manually toggled on channels. Chan_rtp.c: MulticastRTP missing refcount without codec option. Chan_rtp.c: Change MulticastRTP nameing to avoid memory leak. Func_frame_trace: Add CLI command to dump frame queue. User Notes: app_dial: Add dial time for progress/ringing.
21.1.026 Jan 2024 12:25 major feature: - logger: Fix linking regression. - Revert "core res_pjsip: Improve topology change handling." - menuselect: Use more specific error message. - res_pjsip_nat: Fix potential use of uninitialized transport details - app_if: Fix faulty EndIf branching. - manager.c: Fix regression due to using wrong free function. - doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories - config_options.c: Fix truncation of option descriptions. - manager.c: Improve clarity of "manager show connected". - make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation. - general: Fix broken links. - MergeApproved.yml: Remove unneeded concurrency - app_dial: Add option "j" to preserve initial stream topology of caller - pbx_config.c: Don't crash when unloading module. - ast_coredumper: Increase reliability - logger.c: Move LOG_GROUP documentation to dedicated XML file. - res_odbc.c: Allow concurrent access to request odbc connections - res_pjsip_header_funcs.c: Check URI parameter length before copying. - config.c: Log #exec include failures. - make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS. - app_voicemail.c: Completely resequence mailbox folders. - sig_analog: Fix channel leak when mwimonitor is enabled. - res_rtp_asterisk.c: Update for OpenSSL 3+. - alembic: Update list of TLS methods available on ps_transports. - func_channel: Expose previously unsettable options. - app.c: Allow ampersands in playback lists to be escaped. - uri.c: Simplify ast_uri_make_host_with_port() - func_curl.c: Remove CURLOPT() plaintext documentation. - res_http_websocket.c: Set hostname on client for certificate validation. - live_ast: Add astcachedir to generated asterisk.conf. - SECURITY.md: Update with correct documentation URL - func_lock: Add missing see-also refs to documentation. - app_followme.c: Grab reference on nativeformats before using it - configs: Improve documentation for bandwidth in iax.conf. - logger: Add channel-based filtering. - chan_iax2.c: Don'