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.
 Homepage
 Download
Recent Releases
21.6.030 Nov 2024 17:05
minor feature:
Commits: 39.
Commit Authors: 9.
Resolved: 22.
Security Advisories Resolved: 0.
21.5.027 Nov 2024 10:05
minor feature:
Commits: 24.
Commit Authors: 8.
Resolved: 17.
Security Advisories Resolved: 0.
21.4.323 Nov 2024 08:45
minor feature:
Commits: 1.
Commit Authors: 1.
Resolved: 0.
Security Advisories Resolved: 1.
GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming SIP request can cause Asterisk to crash when res_resolver_unbound is used.
21.4.219 Sep 2024 14:05
minor feature:
Commits: 1.
Commit Authors: 1.
Resolved: 0.
Security Advisories Resolved: 1.
GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code execution / System() dialplan.
21.4.118 Aug 2024 12:05
minor feature:
Commits: 2.
Commit Authors: 1.
Resolved: 2.
Security Advisories Resolved: 0.
21.4.029 Jul 2024 21:45
minor feature:
Commits: 20.
Commit Authors: 9.
Resolved: 8.
Security Advisories Resolved: 0.
21.3.126 May 2024 09:05
minor feature:
Commits: 1.
Commit Authors: 1.
Resolved: 0.
Security Advisories Resolved: 1.
GHSA-qqxj-v78h-hrf9: res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests.
21.3.017 May 2024 12:25
minor feature:
Commits: 43.
Commit Authors: 15.
Resolved: 26.
Security Advisories Resolved: 0.
21.2.020 Mar 2024 05:05
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README.md: Removed outdated link.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.
21.1.026 Jan 2024 03:45
minor feature:
21.0.209 Jan 2024 19:42
minor feature:
Links:
----------------------------------------
- Full ChangeLog (https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- GitHub Diff (https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- Tarball (https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- Downloads (https://downloads.asterisk.org/pub/telephony/asterisk)
Summary:
----------------------------------------
- res_rtp_asterisk: Fix regression issues with DTLS client check
User Notes:
----------------------------------------
Upgrade Notes:
----------------------------------------
Closed Issues:
----------------------------------------
- #500: bug regression : res_rtp_asterisk doesn't build if pjproject isn't used
- #503: bug : The res_rtp_asterisk DTLS check against ICE candidates fails when it shouldn't
- #505: bug : res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs created by ast_sockaddr_from_pj_sockaddr()
Commits By Author:
----------------------------------------
- ### George Joseph (1):
- res_rtp_asterisk: Fix regression issues with DTLS client check
Detail:
----------------------------------------
- ### res_rtp_asterisk: Fix regression issues with DTLS client check
Author: George Joseph
Date: 2023-12-15
* Since ICE candidates are used for the check and pjproject is
required to use ICE, res_rtp_asterisk was failing to compile
when pjproject wasn't available. The check is now wrapped
with an #ifdef HAVE_PJPROJECT.
* The rtp- ice_active_remote_candidates container was being
used to check the address on incoming packets but that
container doesn't contain peer reflexive candidates discovered
during negotiation. This was causing the check to fail
where it shouldn't. We now check against pjproject's
real_ice- rcand array which will contain those candidates.
* Also fixed a bug in ast_sockaddr_from_pj_sockaddr() where
we weren't zeroing out
| 
