The release of Asterisk 16.17.0 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:
-----------------------------------
| ASTERISK-29203 / AST-2021-002 -- Another scenario is causing a crash | (Reported by Gregory Massel) | |
| sRTP Replay Protection ignored; even tears down long calls | (Reported by Alexander Traud) | |
| res_pjsip_diversion: sending multiple 181 responses causes memory corruption and crash | (Reported by Ivan Poddubny) |
Bugs fixed in this release:
-----------------------------------
| res_pjsip_session: NULL active_media_state topology caused asterisk crash | (Reported by sungtae kim) | |
| chan_local: Multistream support breaks T.38 faxing | (Reported by Matthias Hensler) | |
| app_confbridge: Memory rises when jitterbuffer enabled and muting over AMI occurs | (Reported by Stefan Ruf) | |
| app_dial: DTMF to 'D' option gets duplicated if there are multiple progress events | (Reported by N A) | |
| Fix differing usage of assignment operators in modules.conf | (Reported by Rusty Newton) | |
| strings: Incorrect use of __attribute__((pure)) in ast_str_to_lower definition | (Reported by Vitezslav Novy) | |
| res_rtp_asterisk: When native local bridging the remote SSRC becomes permanent | (Reported by Sebastian Damm) | |
| res_pjsip_nat: Contact is rewritten on REGISTER responses with external_signaling_address | (Reported by Brian Paboojian) | |
| ICE Role conflict with an unauthorized session | (Reported by Salah Ahmed) | |
| chan_pjsip: 180 Ringing with SDP not changed into progress | (Reported by Sebastian Damm) | |
| say: Y2021 problem â?? Asterisk cannot say year 2021 in Dutch | (Reported by Jacek Konieczny) | |
| res_fax: asterisk fails to publish the Stasis and ReceiveFax status messages if the remote Station ID contains invalid UTF-8 characters | (Reported by Alexei Gradinari) | |
| Callee declined when 'beep' audio file does not exist | (Reported by IAMJames_) | |
| res_pjsip_refer: Segfault in progress notify | (Reported by George Joseph) | |
| pjsip: | (Reported by Michael Maier) | |
| pjsip: Re-invite occurs when it shouldn't | (Reported by Benjamin Keith Ford) | |
| res_config_pgsql: Limit realtime_pgsql() to return one (no more) record | (Reported by Boris P. Korzun) | |
| res_odbc_transaction sets forcecommit default value based on isolation level instead of forcecommit | (Reported by Jaco Kroon) | |
| app_queue: Member device state "invalid" when second call is ringing and hint is used | (Reported by Boolah ) | |
| app.h: C++ compatibility broken | (Reported by Jean Aunis - Prescom) | |
| res_pjsip_t38: Crash when changing state | (Reported by Gregory Massel) | |
| res_rtp_asterisk: Asterisk crashes when making hold/unhold from webrtc client | (Reported by Edvin Vidmar) | |
| res_pjsip: Segmentation fault | (Reported by Mauri de Souza Meneguzzo (3CPlus)) | |
| chan_sip: Allow peers without audio (text+video). | (Reported by Alexander Traud) | |
| chan_sip: Allow text+video media streams, again. | (Reported by Alexander Traud) | |
| channel: Allow text+video media streams, again. | (Reported by Alexander Traud) | |
| res_pjsip: user=phone validation fail for isup numbers containing *# | (Reported by Mark Petersen) | |
| chan_sip: Audio stream rejected, Other stream present: Invalid SDP. | (Reported by Alexander Traud) | |
| After T38 reinvite response of 488 a subsequent G711 reinvite is not processed correctly. Instead the previous T38 session media is used | (Reported by Robert Cripps) | |
| res_pjsip_session: res sometimes uninitialized reported by compiler Clang. | (Reported by Alexander Traud) |
Improvements made in this release:
-----------------------------------
| sorcery: Add support for more intelligent reloading. | (Reported by Joshua C. Colp) | |
| res_pjsip_registrar: Include source IP address and port in log messages | (Reported by Joshua C. Colp) | |
| asterisk: Update copyright/company | (Reported by Joshua C. Colp) | |
| Add MixMonitorStart / Stop / Mute AMI events | (Reported by Sébastien Duthil) | |
| Support of MIME-type for wav16 | (Reported by Boris P. Korzun) | |
| TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER) failure SIP code | (Reported by Dan Cropp) | |
| Support of various URL-schemes by MoH | (Reported by Boris P. Korzun) |
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
Thank you for your continued support of Asterisk!
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