The release of Asterisk 16.1.0 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:
-----------------------------------
| Buffer overflow for DNS SRV/NAPTR records | (Reported by Jan Hoffmann) | |
| res_http_websocket: Crash when reading HTTP Upgrade requests | (Reported by Sean Bright) |
New Features made in this release:
-----------------------------------
| add flag to allow CALLERID(num) to be placed in Contact header in chan_pjsip | (Reported by Torrey Searle) |
Bugs fixed in this release:
-----------------------------------
| app_voicemail: MWI fails with mailboxes=##@device instead of mailboxes=##@default | (Reported by Ronald Raikes) | |
| app_queue: Revert broken queue channel reference patch | (Reported by lvl) | |
| [patch] need to reset DTMF last sequence number and timestamp on voice packet with marker bit | (Reported by Alexei Gradinari) | |
| SIGABRT caused by stack corruption in hashkeys_read when no matching keys present | (Reported by Michael Walton) | |
| repeated segmentation faults | (Reported by Eyal Hasson) | |
| ARI /channels/create handler causes core dump | (Reported by sungtae kim) | |
| stasis: Filter messages at publishing to reduce work done | (Reported by Joshua C. Colp) | |
| Incorrect Behavior for rewrite_contact when Re-Invite omits routset | (Reported by Torrey Searle) | |
| Some conditions prevent running of el_end, break the terminal. | (Reported by Corey Farrell) | |
| rtp: Incorrect Packetization | (Reported by Robert Cripps) | |
| pbx_config: Only the first [globals] section is processed. | (Reported by Corey Farrell) | |
| Formatting error in documentation | (Reported by Scott Griepentrog) | |
| chan_sip: Asterisk 12+ chan_sip doesn't report AST_CEL_PICKUP in handle_invite_replaces | (Reported by Luit van Drongelen) | |
| res_pjsip_notify: improve realtime performance on CLI completion on the endpoint | (Reported by Alexei Gradinari) | |
| Caller ID cannot be changed on Attended Transfer before dialing out | (Reported by Alexei Gradinari) | |
| app_confbridge: Participant info labels aren't being added to the SDPs | (Reported by George Joseph) | |
| function ast_sendtext() create RTP realtime packets with a trailing null byte in the payload | (Reported by Emmanuel BUU) | |
| bridging: Asterisk crashes when receiving an empty realtime text frame | (Reported by Emmanuel BUU) | |
| app_queue: QueueMemberStatus Event flooding AMI | (Reported by Andrej) | |
| res_pjsip: improve realtime performance on CLI 'pjsip show contacts' | (Reported by Alexei Gradinari) | |
| app_queue: Queue member considered inuse after immediately hanging up during dialing. | (Reported by Cao Minh Hiep) | |
| stasis: Playing MOH to bridge with ARI does not work | (Reported by Cameron) | |
| res_odbc: missing SQL error diagnostic | (Reported by Alexei Gradinari) | |
| chan_sip: SipNotify via AMI behaves differently to CLI | (Reported by Peter Katzmann) | |
| configure script does not enforce libunbound2 version | (Reported by Samuel Galarneau) | |
| testsuite: Sniffer assumes pjmedia will use ports below 10000 | (Reported by Joshua C. Colp) | |
| rtp: Crash in off-nominal case where RTP instance can't be set up | (Reported by Lei Fu) | |
| chan_sip unstable with TLS after asterisk start or reloads | (Reported by David Hajek) | |
| PJSIP: Update bundled PJPROJECT to version 2.8 | (Reported by Joshua C. Colp) | |
| res_pjsip_mwi: Memory leak on reload | (Reported by Sergej Kasumovic) | |
| chan_pjsip: Declined video stream is added when no video codecs configured and session refresh with removed video stream occurs | (Reported by Will) | |
| AMI event "NewExten" is set to the wrong class | (Reported by lvl) | |
| res_pjproject build failure | (Reported by Jaco Kroon) | |
| [patch] res_musiconhold : music on hold will not start if previous hold just reached end of file | (Reported by Frederic LE FOLL) | |
| channel.c: ARI ring only once | (Reported by Hajek Michal) | |
| Realtime queuemembers are not updated during retry phase | (Reported by lvl) | |
| alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is integer not boolean | (Reported by Joshua C. Colp) | |
| res_pjsip_transport_websocket: Properly set 'received' for IPv6 | (Reported by Sean Bright) | |
| When T.140 realtime text is negociated, a lot of debug traces are generated | (Reported by Emmanuel BUU) | |
| PBX calls via chan_sip TCP trunk now get authentification error | (Reported by Ian Gilmour) | |
| res_pjsip realtime: uri column in ps_contacts table can be too short | (Reported by Florian Floimair) | |
| res_pjsip_t38: Crash receiving 1xx responses other than 100 before 200 for T.38 reINVITE | (Reported by Joshua Elson) | |
| rtcp-mux is put in SDP answer regardless of offer | (Reported by Torrey Searle) | |
| No joint capabilities with video and audio-only streams | (Reported by Benjamin Keith Ford) | |
| app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY | (Reported by Valentin Safonov) | |
| pjproject_bundled: Fix for Solaris builds. Do not undef s_addr. | (Reported by Alexander Traud) | |
| Wrong SRTP use status report | (Reported by Salah Ahmed) | |
| res_pjsip_registrar: Improve performance of inbound handling | (Reported by Joshua C. Colp) | |
| pjsip: Race condition in 183 re transmission can result in a deadlock | (Reported by Torrey Searle) | |
| make menuselect fails due to undefined symbols (initscr32, w32addch) in menuselect_curses.o | (Reported by Majdi Bsoul) | |
| [regression] menuselect compilation failure on Solaris 10 | (Reported by Samuel Owens) | |
| menuselect compilation failure on Solaris 10 / gcc 3.4.3 | (Reported by rleasure) | |
| menuselect compilation failure on Solaris 10/gcc-4.1.1 | (Reported by Bob Atkins) | |
| BuildSystem: Enable Jansson in Solaris 11. | (Reported by Alexander Traud) | |
| res_pjsip_endpoint_identifier_ip only matches against "generic string" headers | (Reported by George Joseph) | |
| res_rtp_asterisk: Requires OpenSSL in Developer Mode. | (Reported by Alexander Traud) | |
| Frack errors in stasis.c and memory leakage | (Reported by Siruja Maharjan) | |
| res_pjsip: Change default transport keepalive to preserve behavior | (Reported by Joshua C. Colp) | |
| systemd: asterisk.service | (Reported by seanchann.zhou) |
Improvements made in this release:
-----------------------------------
| [patch] New function PJSIP_PARSE_URI to parse an URI and return a specified part of the URI | (Reported by Alexei Gradinari) | |
| Allow the sip_to_pjsip script to be used in a pipe | (Reported by Pascal Cadotte Michaud) | |
| Remove stale nonoptreq references | (Reported by Walter Doekes) | |
| [patch] Add IPv6 Support for DUNDi | (Reported by Adam Secombe) | |
| PJSIP: Missing "party=calling"/"party=called" in Remote-Party-ID | (Reported by Eric Dantie) | |
| pjproject_bundled: Find shared libraries in root --with-ssl=PATH. | (Reported by Alexander Traud) | |
| pjsip_wizard example gives wrong info about unsupported SRV records | (Reported by Jonathan Harris) | |
| res_rtp_asterisk: T.140 packets containing backspace or end of line are merged with regular text and it causes some UA to break | (Reported by Emmanuel BUU) |
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
Thank you for your continued support of Asterisk!
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