The release of Asterisk 16.0.0 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:
-----------------------------------
| res_http_websocket: Crash when reading HTTP Upgrade requests | (Reported by Sean Bright) | |
| iostreams: Potential DoS when client connection closed prematurely | (Reported by Sean Bright) | |
| Username bruteforce is possible when using ACL with PJSIP | (Reported by John) | |
| WebSocket frames with 0 sized payload causes DoS | (Reported by Sean Bright) | |
| Segmentation fault occurs in asterisk with an invalid SDP fmtp attribute | (Reported by Sandro Gauci) | |
| Segmentation fault occurs in Asterisk with an invalid SDP media format description | (Reported by Sandro Gauci) | |
| Crash occurs when sending a repeated number of INVITE messages over TCP or TLS transport | (Reported by Sandro Gauci) | |
| SUBSCRIBE message with a large Accept value causes stack corruption | (Reported by Sandro Gauci) |
New Features made in this release:
-----------------------------------
| Add the ability to read the media file type from HTTP header for playback | (Reported by Gaurav Khurana) | |
| Add cache_pools debug option to pjproject.conf | (Reported by Richard Mudgett) | |
| Add new AMI Action for PJSIPShowContacts | (Reported by sungtae kim) | |
| res_pjsip: Add new AMI Action for PJSIPShowAuths | (Reported by sungtae kim) | |
| core: Add support for timelen parsing to ast_parse_arg and ACO. | (Reported by Corey Farrell) | |
| PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming INVITE Request-URI. | (Reported by Richard Mudgett) | |
| Add cache_media_frames debugging option. | (Reported by Richard Mudgett) | |
| res_pjsip: No mechanism exists to limit endpoint identification to IP only | (Reported by Ben Merrills) | |
| [patch]AMI : Add CancelAtxfer Action | (Reported by Thomas Sevestre) | |
| [New Feature] Add mute and DTMF passthrough to ARI add channel to bridge | (Reported by Darren Sessions) | |
| [patch]chan_sip: Access incoming SIP REFER headers in the dialplan | (Reported by Kirill Katsnelson) | |
| chan_sip: Dialplan function SIP_HEADERS() to complement SIP_HEADER(). | (Reported by Kirill Katsnelson) | |
| Add support for systemd socket activation | (Reported by Corey Farrell) |
Bugs fixed in this release:
-----------------------------------
| AMI event "NewExten" is set to the wrong class | (Reported by lvl) | |
| alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is integer not boolean | (Reported by Joshua C. Colp) | |
| res_pjsip realtime: uri column in ps_contacts table can be too short | (Reported by Florian Floimair) | |
| res_pjsip: Change default transport keepalive to preserve behavior | (Reported by Joshua C. Colp) | |
| [patch] pjproject_bundled: Repair ./configure --with-ssl=PATH. | (Reported by Alexander Traud) | |
| BASIC-RETRANS: Implement receive | (Reported by Benjamin Keith Ford) | |
| res_sorcery_config: Allow object name based matching | (Reported by Joshua C. Colp) | |
| module: Remove old modules, update support levels | (Reported by Joshua C. Colp) | |
| stasis: Improve message type "Use of before init/after destruction" error | (Reported by Joshua C. Colp) | |
| srtp: rejecting short sdes lifetimes incompatible with obihai ATAs | (Reported by Nick French) | |
| res_pjsip: Spurious ERROR logging when printing headers in sip_msg | (Reported by Nick French) | |
| pjsip modules always get -O2 even when DONT_OPTIMIZE is set | (Reported by George Joseph) | |
| [patch] pjproject_bundled: Disable TCP/TLS keep-alives. | (Reported by Alexander Traud) | |
| PJSIP proposes ICE candidates on answer even if not in offer | (Reported by Torrey Searle) | |
| [patch] Compile fails with `IPTOS_MINCOST' undeclared. | (Reported by Alexander Traud) | |
| res_pjsip_session: sdp group:BUNDLE attribute truncated | (Reported by Kevin Harwell) | |
| res_pjsip_pubsub: segfault in function publish_expire | (Reported by Alexei Gradinari) | |
| res_pjsip_rfc3326: A lot of endpoints do not correctly handle two Reason headers | (Reported by Ross Beer) | |
| res_pjsip_session: Initial INVITE with audio+fax results in 488 instead of declining stream | (Reported by Thiago Coutinho) | |
| res_pjsip_t38: ATA fails with hangupcause 58(Bearer capability not available) | (Reported by Jared Hull) | |
| res_pjsip_t38: Slow T.38 re-invite rejection if remote leg has T.38 disabled | (Reported by Torrey Searle) | |
| res_pjsip: Lock inversion in transport management | (Reported by Ross Beer) | |
| [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11. | (Reported by Alexander Traud) | |
| res_pjsip_t38: Crash receiving 1xx responses other than 100 before 200 for T.38 reINVITE | (Reported by Joshua Elson) | |
| res_pjsip_pubsub: apparent crash on shutdown | (Reported by Kevin Harwell) | |
| app_confbridge: Conference bridge and announcer channels are not removed if conference is ended as soon as it starts | (Reported by Robert Mordec) | |
| cdr: Deadlock with submit_scheduled_batch and submit_unscheduled_batch | (Reported by Denis Lebedev) | |
| pbx_dundi: Asterisk crashes when unloading module pbx_dundi.so with dundi peers | (Reported by Kirsty Tyerman) | |
| AMI: Action SendText needs to use the correct thread. | (Reported by Richard Mudgett) | |
| res_pjsip_messaging doesn't accept application/* content-types. | (Reported by George Joseph) | |
| res_pjsip_session doesn't update media when a 200 comes in with a different port than a 183 | (Reported by George Joseph) | |
| [patch] uuid: Enable UUID in Solaris 11. | (Reported by Alexander Traud) | |
| channels: CHECK_BLOCKING is ineffective | (Reported by Corey Farrell) | |
| [patch] BuildSystem: Enable ./configure in Solaris 11. | (Reported by Alexander Traud) | |
| [patch] bootstrap.sh: find -maxdepth is not POSIX compatible. | (Reported by Alexander Traud) | |
| menuselect: GCC 8: restrict-qualified parameter passed and aliased. | (Reported by Alexander Traud) | |
| [patch] tests/test_utils: Repair ./configure --with-ssl=PATH. | (Reported by Alexander Traud) | |
| chan_iax2: Stops listening for traffic | (Reported by Kirsty Tyerman) | |
| rtp: DTMF Breaks With telephony-event/16000 | (Reported by Dominic) | |
| [patch] crypto.h: Repair ./configure --with-ssl=PATH. | (Reported by Alexander Traud) | |
| [patch] res_srtp: Repair ./configure --with-ssl=PATH. | (Reported by Alexander Traud) | |
| SQL fetch error on query which return 0 columns | (Reported by Alexei Gradinari) | |
| chan_pjsip isn't updating hangupcause on 4XX responses | (Reported by George Joseph) | |
| [patch] ooh323c: GCC 8: output truncated before terminating nul. | (Reported by Alexander Traud) | |
| res_pjsip: Modified qualify_frequency doesn't effect until pjsip reload | (Reported by Alexei Gradinari) | |
| res_fax: Deadlock when using Local channels and fax gateway | (Reported by David Brillert) | |
| Manager events for MeetMe have incorrectly documented key name 'Usernum' - should be 'User' | (Reported by Francois Blackburn) | |
| [patch] tcptls.h: Repair ./configure --with-ssl=PATH. | (Reported by Alexander Traud) | |
| [patch] tcptls: Allow OpenSSL configured with no-dh. | (Reported by Alexander Traud) | |
| [patch] tcptls: Allow OpenSSL 1.1.x configured with enable-ssl3-method no-deprecated. | (Reported by Alexander Traud) | |
| Codec-Change Re-INVITE during DTMF can cause marker bit error | (Reported by Torrey Searle) | |
| res_rtp_asterisk: Add support for abs-send-time RTP extension | (Reported by Joshua C. Colp) | |
| config/ast_destroy_realtime_fields: successful DELETE is treated as failed | (Reported by Alexei Gradinari) | |
| [patch]: tcptls: Repair ./configure --with-ssl=PATH. | (Reported by Alexander Traud) | |
| Asterisk ODBC Voicemail Prompt storage fails with recent MariaDB version. | (Reported by Nic Colledge) | |
| Incorrect error reported when leaving/retrieving a ODBC voicemail | (Reported by Nic Colledge) | |
| chan_mobile: presents incorrect inbound Caller-ID names | (Reported by Brian) | |
| [patch] res_pjsip_endpoint_identifier_ip: Unregister the module for headers. | (Reported by Alexander Traud) | |
| cli: "manager show settings" mislabels HTTP timeout as being minutes. | (Reported by Corey Farrell) | |
| Fix issues exposed by GCC 8 | (Reported by George Joseph) | |
| [patch] rtp_engine: Allow Media Formats with add_static_payload(-1) on egress again. | (Reported by Alexander Traud) | |
| [patch] sip_to_pjsip: Enable python3 compatibility. | (Reported by Alexander Traud) | |
| digest over for manager (ami) over http fails on too long uris | (Reported by Jaco Kroon) | |
| Macro allows an infinite loop of dialplan inclusion resulting in a crash | (Reported by Tzafrir Cohen) | |
| cdr_mysql creates empty records if reconnects when mysql was not up on module load | (Reported by Tzafrir Cohen) | |
| Asterisk got stuck while enabling "ari set debug all on" | (Reported by shaurya jain) | |
| chan_sip: one way / no audio with srtp | (Reported by Florian Kaiser) | |
| One way audio when calling from Asterisk(sip trunk) to another number where both are connected to a SBC using TLS+SRTP | (Reported by Artur Pires) | |
| pjsip_options: rework to make more efficient | (Reported by Kevin Harwell) | |
| translate: interpolated frames are not passed through | (Reported by Kevin Harwell) | |
| When the ooh323 debug is on there is no ringing signal to incoming calls via H323 trunk. | (Reported by Dimos) | |
| No "alert" or "progress" in chan_ooh323 if debug is enabled only on the module | (Reported by Marco Giordani) | |
| bridge_softmix / app_confbridge: Add support for combining REMB reports | (Reported by Joshua C. Colp) | |
| [patch] BuildSystem: Enable IMAP storage on FreeBSD and DragonFly BSD. | (Reported by Alexander Traud) | |
| app_confbridge: "core show profile bridge" does not output "sfu" when video_mode is sfu | (Reported by Carlos Chavez) | |
| [patch] utils/pval: Add -lBlocksRuntime for compiler clang conditionally. | (Reported by Alexander Traud) | |
| [patch] chan_vpb: Avoid GNU old-style field designator extension. | (Reported by Alexander Traud) | |
| BASIC-RETRANS: Implement send | (Reported by Benjamin Keith Ford) | |
| res_musiconhold: Music on hold restarts after every announcement | (Reported by lvl) | |
| cdr_mysql: Missing MYSQL_PORT definition | (Reported by Evandro César Arruda) | |
| res_pjsip_session: SDP origin does not use resolved address | (Reported by John M.) | |
| res_rtp_asterisk: Add support for sending RTCP feedback messages | (Reported by Joshua C. Colp) | |
| chan_sip: New Channel creation from new SIP dialog with Replaces failed to be properly tracked and destroyed | (Reported by Shannon Price) | |
| app_confbridge: Add ability to enable and configure REMB support | (Reported by Joshua C. Colp) | |
| PJSIP: Deadlock shutting down subscription TCP connection and sending subscription message. | (Reported by Ross Beer) | |
| res_pjsip: Crash on TCP PJSIP Transport Disconnect | (Reported by Ross Beer) | |
| res_rtp_asterisk: Add support for raising RTCP feedback messages | (Reported by Joshua C. Colp) | |
| rtp: RTCP messages with REMB trigger fast picture update | (Reported by Joshua C. Colp) | |
| Command line not being parsed correctly with getopt not from glibc | (Reported by Guido Falsi) | |
| [patch] configure: pjsip_evsub_set_uas_timeout not found. | (Reported by Alexander Traud) | |
| [patch] BuildSystem: With external editline, do not require libs for internal editline. | (Reported by Alexander Traud) | |
| ConfBridge: raise ConfbridgeTalking when put on hold and clear talking status | (Reported by Kevin Harwell) | |
| Generic PLC doesn't work if the 2 codecs on a channel are equal | (Reported by George Joseph) | |
| [patch] BuildSystem: Remove unused dependency on libltdl. | (Reported by Alexander Traud) | |
| [patch] Make format_ogg_vorbis work on OpenBSD | (Reported by Michiel van Baak) | |
| [patch] BuildSystem: Enable Advanced Linux Sound Architecture (ALSA) in NetBSD. | (Reported by Alexander Traud) | |
| res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't account for more than one 'Reason' header | (Reported by Ross Beer) | |
| [patch] BuildSystem: Enable IMAP storage on openSUSE and Arch Linux. | (Reported by Alexander Traud) | |
| [patch] install_prereq: Update FreeBSD libraries. | (Reported by Alexander Traud) | |
| [patch] res_srtp: Add support for libsrtp2.x on openSUSE. | (Reported by Alexander Traud) | |
| NetBSD Build Needs RPATH set in 1.2.25 | (Reported by Curt Sampson) | |
| BuildSystem: Enable Better Backtraces in FreeBSD. | (Reported by Alexander Traud) | |
| Deprecate legacy modules | (Reported by Corey Farrell) | |
| uuid_generate_random detection failure | (Reported by John Nemeth) | |
| [patch] BuildSystem: Enable PortAudio in NetBSD. | (Reported by Alexander Traud) | |
| [patch] BuildSystem: AC_PATH_PROG sets to colon character when not found. | (Reported by Alexander Traud) | |
| res_pjsip_rfc3326: Order of 'Reason' headers break many endpoints | (Reported by Ross Beer) | |
| AMI Action VoicemailUsersList returns 0 MessageCount | (Reported by Sébastien Duthil) | |
| chan_sip: RTP framing issues on outgoing calls | (Reported by Jean Aunis - Prescom) | |
| PJSIP: Forked INVITE SDP negotiation gets one way audio. | (Reported by lvl) | |
| [patch] BuildSystem: Enable Lua in NetBSD. | (Reported by Alexander Traud) | |
| [patch] BuildSystem: Depend not implicitly but explicitly on external libraries. | (Reported by Alexander Traud) | |
| [patch] res_http_post: Enable GMime in NetBSD. | (Reported by Alexander Traud) | |
| [patch] BuildSystem: Enable autotools in NetBSD. | (Reported by Alexander Traud) | |
| [patch] chan_unistim: NetBSD has an incompatible struct in_pktinfo. | (Reported by Alexander Traud) | |
| [patch] BuildSystem: Cast any intptr_t explicitly to its proposed type. | (Reported by Alexander Traud) | |
| [patch] BuildSystem: Detect whether uselocale(.) is available. | (Reported by Alexander Traud) | |
| [patch] BuildSystem: Avoid re-defining of pthread_* on NetBSD. | (Reported by Alexander Traud) | |
| [patch] BuildSystem: Install init scripts on openSUSE Tumbleweed. | (Reported by Alexander Traud) | |
| [patch] BuildSystem: Avoid == for comparison in ./configure. | (Reported by Alexander Traud) | |
| app_amd.so returning TOOLONG before reaching the timeout | (Reported by Michael Cargile) | |
| Documentation: voicemail.conf.sample shows 512 limit for emailbody field, however this is only true if compiled with LOW_MEMORY option | (Reported by Fran Vicente) | |
| PJSIP: Crash during SIP attended transfer. | (Reported by Bryan Walters) | |
| Output from rawman truncated if output is long enough | (Reported by Bojan NemÄ?iÄ?) | |
| bridging: Sometimes cloning the stream topology causes a crash | (Reported by Richard Mudgett) | |
| core: If frame with unnegotiated format is read crash will occur | (Reported by Sébastien Duthil) | |
| Wrong remote identity and target in dialog package XML in NOTIFY | (Reported by Alejandro Padilla) | |
| Asterisk "doc/lang/language-criteria.txt" needs update or removal. | (Reported by Rusty Newton) | |
| ICE fails with no candidate nominated | (Reported by Thomas Guebels) | |
| [patch] rtp_engine: Load format name / mime type in uppercase again. | (Reported by Alexander Traud) | |
| res_pjsip: Endpoint destruction does not free DTLS configuration | (Reported by Mak Dee) | |
| [patch] install_prereq: Update OpenBSD libraries. | (Reported by Alexander Traud) | |
| [patch] res_calendar: Specialized calendars depend on symbols of general calendar. | (Reported by Alexander Traud) | |
| [patch] BuildSystem: Enable IMAP storage on OpenBSD. | (Reported by Alexander Traud) | |
| [patch] BuildSystem: Enable system provided libedit on OpenBSD. | (Reported by Alexander Traud) | |
| [patch] BuildSystem: Remove chan_h323 leftovers. | (Reported by Alexander Traud) | |
| [patch] BuildSystem: Invoke ldconfig with previous paths. | (Reported by Alexander Traud) | |
| [patch] BuildSystem: Do not warn when bash is not installed. | (Reported by Alexander Traud) | |
| chan_sip: Crash processing CANCEL request | (Reported by Leandro Dardini) | |
| Internal pjproject build doesn't disable bcg729 | (Reported by Stuart Henderson) | |
| [patch] codecs: Add support for WebRTC iLBC 2.0. | (Reported by Alexander Traud) | |
| Determine if the internal editline and stdtime libraries are still relevant | (Reported by George Joseph) | |
| [patch] backtrace: Avoid -Wlogical-not-parentheses. | (Reported by Alexander Traud) | |
| [patch] install_prereq: Update Debian/Ubuntu libraries. | (Reported by Alexander Traud) | |
| CDR: Leaking channel snapshots allocated by stasis_channel.c | (Reported by Kristijan Vrban) | |
| chan_console: cannot read and write at the same time with alsa backend | (Reported by Tzafrir Cohen) | |
| (null) string tailing after AsyncAGIEnd AMI event | (Reported by sungtae kim) | |
| Null pointer Crash in PJSIP MWI | (Reported by Joshua Elson) | |
| res_pjsip: If SIP response is received during shutdown a crash may occur | (Reported by Joshua C. Colp) | |
| Build System: Require compiler to provide built-in support for atomic references. | (Reported by Corey Farrell) | |
| Subscriptions Persist After Expiration and TCP/TLS Disconnect | (Reported by Ross Beer) | |
| [patch] BuildSystem: Enable autotools in FreeBSD. | (Reported by Alexander Traud) | |
| [patch] app_voicemail: Avoid always true warnings with clang. | (Reported by Alexander Traud) | |
| [patch] install_prereq: Update RHEL/CentOS/Fedora libraries. | (Reported by Alexander Traud) | |
| core: macOS devmode build fails: variable 'freeswap' set but not used | (Reported by David M. Lee) | |
| [patch] editline: Avoid shifting a negative signed value. | (Reported by Alexander Traud) | |
| Problems with siren14 codec; problems with siren7 sound files. | (Reported by Steve Murphy) | |
| [patch] configure.ac in 1.4.37 broken with autoconf 2.60 | (Reported by Stéphan Kochen) | |
| [patch] install_prereq: Download latest Jansson. | (Reported by Alexander Traud) | |
| New module loader aborts startup if a required module declines load. | (Reported by snuffy) | |
| [patch] res_config_mysql: Avoid the header mysql_version.h. | (Reported by Alexander Traud) | |
| When running ./contrib/scripts/install_prereq install-unpackaged pjproject is installed in wrong place | (Reported by PowerPBX) | |
| [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory. | (Reported by Alexander Traud) | |
| [patch] BuildSystem: Allow make clean all again. | (Reported by Alexander Traud) | |
| [patch] install_prereq: Support package manager DNF. | (Reported by Alexander Traud) | |
| Placing call on hold temporarily locks up set | (Reported by Igor Goncharovsky) | |
| [patch] BuildSystem: Use the detected name for MD5 everywhere. | (Reported by Alexander Traud) | |
| [patch] BuildSystem: Invoke install not in GNU but POSIX style. | (Reported by Alexander Traud) | |
| [patch] BuildSystem: In OpenBSD, xmlstarlet is xml. | (Reported by Alexander Traud) | |
| [patch] BuildSystem: Detect external library Lua in version 5.3. | (Reported by Alexander Traud) | |
| res_pjsip_endpoint_identifier_ip only matches against header if match by ip fails | (Reported by George Joseph) | |
| res_pjsip: Segfault when calling pjsip_hdr_print_on in sip_msg.c:581 | (Reported by Ross Beer) | |
| [patch] BuildSystem: Avoid $EUID and use id -u instead. | (Reported by Alexander Traud) | |
| [patch] BuildSystem: Resolve resolv.h not via Generic but Particular Header-Check. | (Reported by Alexander Traud) | |
| menuselect : remove obsolete TRACE_FRAMES compiler flag | (Reported by Jean Aunis - Prescom) | |
| [patch] res_config_pgsql: Avoid typecasting an int to unsigned char. | (Reported by Alexander Traud) | |
| [patch] clang 5 does not know -Wno-format-truncation | (Reported by Alexander Traud) | |
| [patch] app_osplookup.c: Avoid a format truncation. | (Reported by Alexander Traud) | |
| [patch] chan_ooh323: Avoid typecasting an int to unsigned short. | (Reported by Alexander Traud) | |
| chan_sip: Assumes iostream is non-NULL when it may not be | (Reported by Lubos Dolezel) | |
| [patch] translate: Avoid absolute value on unsigned substraction. | (Reported by Alexander Traud) | |
| res_pjsip_session: Improve WebRTC interop with bundling during renegotiation | (Reported by Joshua C. Colp) | |
| [patch] res_curl: Avoid error message on unload. | (Reported by Alexander Traud) | |
| [patch] clang 5.0: implicit conversion to char changes value to negative. | (Reported by Alexander Traud) | |
| [patch] bridge_softmix: Avoid warning about an uninitialized variable. | (Reported by Alexander Traud) | |
| [patch] editline: Avoid comparison between pointer and zero character constant. | (Reported by Alexander Traud) | |
| [patch] codec_gsm: Avoid shifting a negative signed value. | (Reported by Alexander Traud) | |
| Asterisk configure fails on 'cannot find ptlib-config', despite ptlib-config existing | (Reported by Rusty Newton) | |
| [patch] chan_ooh323: Limit outgoinglimit to positive values as intended. | (Reported by Alexander Traud) | |
| [patch] ooh323cDriver: Fix typo in header guard. | (Reported by Alexander Traud) | |
| [patch] Avoid obsolete warnings on autoconf. | (Reported by Alexander Traud) | |
| Modules need to ensure that any functions, apps, AMI actions, etc. they register are unregistered if the module declines loading | (Reported by Mark Michelson) | |
| 'cdr submit' fails: batch mode not enabled. | (Reported by Tzafrir Cohen) | |
| ICE candidate parser - ICE foundation parsing too short | (Reported by Michele Prà ) | |
| Datastore: Implement automatic module references. | (Reported by Corey Farrell) | |
| Asterisk Turkish Language Set Problem | (Reported by Halil Ä°brahim YILDIZ) | |
| Documentation fix - MASTER_CHANNEL Unexpected Behaviour | (Reported by Shane Mitchell) | |
| Compiler optimizations can break module load sequence. | (Reported by abelbeck) | |
| Security: Authenticated SUBSCRIBE without Contact crashes asterisk | (Reported by Ross Beer) | |
| Typo's | (Reported by Walter Doekes) | |
| bridge: Old channel video source not set to NULL after unref | (Reported by Richard Kenner) | |
| DNS: Unexpected rr_type can cause crash | (Reported by Corey Farrell) | |
| AMI bridge of channels results in MOH not destroyed and robotic audio on one channel | (Reported by Zane Conkle) | |
| chan_console: 'set active' fails to work | (Reported by Tzafrir Cohen) | |
| Asterisk Hangs with Bad file descriptor on read() | (Reported by Abhay Gupta) | |
| ConfBridge sound_muted does not work from CLI or AMI | (Reported by Thomas Frederiksen) | |
| Transfer application does not work with Local channels - documentation misleading | (Reported by Ivan Ullmann) | |
| chan_sip: "rejected because extension not found" should be logged as a security event | (Reported by Brian J. Murrell) | |
| Strictrtp has issues to qualify video rtp streams | (Reported by Wim De Vlaminck) | |
| Coverity Report: Fix issues for error type CHAR_IO | (Reported by Matt Jordan) | |
| iax.conf demo peer is invalid | (Reported by Tzafrir Cohen) | |
| README refers to security documents that do not exist. | (Reported by Corey Farrell) | |
| "core set verbose" behaves strangely, can't alias it, cli.conf example broken | (Reported by Tim Ringenbach at Asteria Solutions Group) | |
| crash after an invalid rtcp packet from GT48 FXS gateway | (Reported by Tzafrir Cohen) | |
| res_rtp_asterisk: Multiple reports in an RTCP packet will write past where it should | (Reported by Vitezslav Novy) | |
| Identify causes and fix pjsip/resolver/srv/failover/in_dialog/transport_tcp | (Reported by Corey Farrell) | |
| Queue members with hints for state_interface get stuck in "In Use" state. | (Reported by Steven T. Wheeler) | |
| chan_sip: Crash Asterisk (in sip_request_call at chan_sip.c) by making a call to a single character in a dot pattern match | (Reported by Dwayne Hubbard) | |
| pjsip_options: qualify_frequency sometimes not applied on reload | (Reported by John Bigelow) | |
| CDR: Deadlock using AMI Originate with Variable CDR(amaflags)=... | (Reported by Richard Mudgett) | |
| RTP: Blind transfer direct media scenario results in one way audio. | (Reported by Richard Mudgett) | |
| SIP ICE support - remove hardcoded limitation on SDP size, make ICE support disabled by default in SIP, maybe provide a better warning message | (Reported by Roy) | |
| chan_sip: Guests disallowed via TCP (or TLS) if existing peer from same IP. | (Reported by Alexander Traud) | |
| pjsip: Clean up WebRTC disables | (Reported by abelbeck) | |
| Security: chan_skinny: Memory exhaustion if flooded with unauthenticated requests | (Reported by George Joseph) | |
| res_http_post: Don't require GMIME_MAJOR_VERSION | (Reported by Joshua C. Colp) | |
| Transcoding makes bad choice in high-rate translations | (Reported by Richard Kenner) | |
| ARI: Updating a bridge gives wrong error message. | (Reported by Frank Durden) | |
| [patch] column and row headers for Signed Linear format variants in output of 'core show translation' are ambiguous | (Reported by Rusty Newton) | |
| H323 audio starts with a delay of 2 seconds. | (Reported by Marco Giordani) | |
| pjsip: 183 without To tag does not negotiate media | (Reported by Kevin Harwell) | |
| [patch] ICE: server-reflexive candidates (srflx) with Dual-Stack. | (Reported by Alexander Traud) | |
| [patch] chan_sip/ICE: Square brackets around IPv6 addresses. | (Reported by Alexander Traud) | |
| Asterisk fails to configure on MacOS Sierra | (Reported by Ivan Larionov) | |
| Asterisk fails to build when openssl headers are not installed. | (Reported by Corey Farrell) | |
| RTP source learning not working with devices that have some clock issues | (Reported by nappsoft) | |
| Attended transfer crashes in Asterisk 13.17.2 | (Reported by Alessandro Pimenta) | |
| Bridging: Crash freeing a frame that's already been freed | (Reported by Richard Kenner) | |
| core: Audiohook freeing interpolated frame when it shouldn't. | (Reported by Mikhail) | |
| app_record: We set the RECORD_STATUS channel variable before closing the file | (Reported by George Joseph) | |
| res_hep_pjsip: For WebRTC clients Asterisk insert same ip address in "source ip address" and "destination ip address" fields in HEP packets | (Reported by Max Norba) | |
| res_http_websocket: Wrong LocalAddress (it is equal to RemoteAddress) | (Reported by Vasilii Rogin) | |
| asterisk.conf: Setting astctl without setting astrundir is ineffective. | (Reported by Corey Farrell) | |
| pjsip: TCP connections may not be destroyed | (Reported by Joshua C. Colp) | |
| DEBUG_FD_LEAKS does not record socketpair, timerfd_create or eventfd. | (Reported by Corey Farrell) | |
| res_pjsip_session: RTP instances leak on 488 responses. | (Reported by Corey Farrell) | |
| chan_sip: Security vulnerability with client code header (revisited) | (Reported by Richard Mudgett) | |
| (Security) Function in PJSIP 2.7 miscalculates the length of an unsigned long variable in 64bit machines | (Reported by Kim youngsung) | |
| Regression: Deadlock between AOR named lock and pjproject grp lock | (Reported by shaurya jain) | |
| res_pjsip: Crash occurs when an empty contact read from astdb or database | (Reported by Aaron An) | |
| res_pjsip: PIDF contact field has malformed/invalid XML | (Reported by basildane) | |
| res_pjsip: TLS options do not handle empty values | (Reported by seanchann.zhou) | |
| srtp: Add support for ephemeral DTLS certificates | (Reported by Sean Bright) | |
| format_ogg_opus: remove from source | (Reported by Kevin Harwell) | |
| [patch] tcptls: Print notice when TLS is enabled but not configured. | (Reported by Alexander Traud) | |
| [patch] libsrtp-2.x.x + AES-GCM support | (Reported by Alexander Traud) | |
| Modules: Fix issues with CLI completion. | (Reported by Corey Farrell) | |
| Regression: pjsip 13.18.0 - from_user - "+" character isn't allowed any more | (Reported by Michael Maier) | |
| channel: Crash when fax gateway is in use with PJSIP | (Reported by Jared Hull) | |
| Audit menuselect module dependencies | (Reported by Corey Farrell) | |
| Optional API modules should not allow unload. | (Reported by Corey Farrell) | |
| Bridge() dialplan application fails without setting BRIDGERESULT channel variable | (Reported by James Terhune) | |
| res_ari_channels: channel_state_invalid always leaks snapshot reference. | (Reported by Marin Odrljin) | |
| stream: Allow streams on a topology to be put into groups | (Reported by Joshua C. Colp) | |
| alembic: PJSIP scripts are missing column bundle in ps_endpoints table | (Reported by Florian Floimair) | |
| Typo in CHANNEL(dtmf_features) usage documentation | (Reported by Igor Goncharovsky) | |
| GCC 7 warning: app_voicemail.c: In function 'imap_delete_old_greeting' | (Reported by Anthony Messina) | |
| jitterbuffer: Does not handle case where translator returns null frame. | (Reported by Joshua Elson) | |
| ARI: Node ARI client broken in latest versions of 13 and 14 | (Reported by Benjamin Keith Ford) | |
| core: Disabling xmldoc support does not work. Also results in abort during Asterisk startup. | (Reported by Mr Dini) | |
| Expires handling in SUBSCRIBE confuses the absence of the Expires header field with an unsubscribe action. | (Reported by Jonathan Cloots) | |
| The config_hook unit test causes Asterisk to crash if run a second time | (Reported by George Joseph) | |
| res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6 set to yes | (Reported by Martin Cisárik) | |
| res_xmpp: Crash if OAuth 2.0 is used before curl is loaded | (Reported by Ronald Raikes) | |
| [patch] chan_sip: Crypto attribute not last but first on SDP media level. | (Reported by Alexander Traud) | |
| res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on un/re-load: mod.id == -1 | (Reported by Tzafrir Cohen) | |
| Cannot disable SIP debugging via CLI after enabling with conf file option - also 'sip set debug off' reports debugging disabled, when it really isn't | (Reported by Rusty Newton) | |
| app_macro deprecation | (Reported by Corey Farrell) | |
| bridge_softmix: When a channel leaves add in any missing participant streams | (Reported by Joshua C. Colp) | |
| sip_to_pjsip not correctly handling disallow=all directive | (Reported by Torrey Searle) | |
| Fails to build in FreeBSD due to sys/sysmacros.h not existing there | (Reported by Guido Falsi) | |
| [patch] res_pjsip_session: SIP/SDP origin (o=) contains local address. | (Reported by Alexander Traud) | |
| chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller | (Reported by lvl) | |
| backtrace.c: Crash due to double-free. | (Reported by Corey Farrell) | |
| [patch] Crash on ast_ssl_teardown when stopping. | (Reported by Alexander Traud) | |
| res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be. | (Reported by dtryba) | |
| res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs | (Reported by dtryba) | |
| [patch] app_queue: Music On Hold for real-time queues is not reset to default | (Reported by Nathan Bruning) | |
| Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate | (Reported by Allen Ford) | |
| cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured | (Reported by Tzafrir Cohen) | |
| Missing openssl dependencies in res_rtp_asterisk and tcptls | (Reported by Tzafrir Cohen) | |
| res_pjsip: Loss of SIP registrations causing unavailable endpoints | (Reported by Richard Mudgett) | |
| res_ari: Memory leaks in ARI when using Content-Type: application/json | (Reported by David Hajek) | |
| chan_sip: tcpbind uses wrong source address | (Reported by Ksenia) | |
| [patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS | (Reported by Alexander Traud) | |
| vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED. | (Reported by Corey Farrell) | |
| res_pjsip_mwi: uninitialized value from ast_strings_match | (Reported by Corey Farrell) | |
| Status of RFC 3323 and PJSIP | (Reported by dtryba) | |
| [patch] False positive busy checks when icalendar's recurrence-id mechanism is involved | (Reported by Benoît Dereck-Tricot) | |
| app_queue: does its check-makeannouncement-logic twice each head-caller-loop | (Reported by Stefan Engström) | |
| Problem with expires on pjsip / outbound-publish | (Reported by Cyrille Demaret) | |
| Contact is improperly translated after d178f497 | (Reported by Sean Bright) | |
| Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes) | (Reported by Ross Beer) | |
| A codeblock that maintains a bug,but maybe the codeblock will never run | (Reported by Huangyx) | |
| bridge: Renegotiate if source stream changes. | (Reported by Joshua C. Colp) | |
| res_pjsip_session: Crashes after sending PRACK and receiving 200 OK | (Reported by Daniel Heckl) | |
| Realtime config fail with PostgreSQL version before 9.1 | (Reported by Rodrigo Ramirez Norambuena) | |
| [pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36 | (Reported by Daniel Heckl) | |
| bridge_native_rtp: half-way direct media when using early bridging | (Reported by Jean Aunis - Prescom) | |
| SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -> 0 | (Reported by Marcello Ceschia) | |
| Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability | (Reported by Ross Beer) | |
| module reload res_calendar.so does not reload everything in calendar.conf | (Reported by Jesper) | |
| RTCP needs better packet validation to resist port scans. | (Reported by Richard Mudgett) | |
| RTP: One way audio with direct media and strictrtp=yes. | (Reported by Richard Mudgett) | |
| res_calendar does not process CalDAV from Owncloud [fix included] | (Reported by Stefan Gofferje) | |
| res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured. | (Reported by Jesper) | |
| RTP Multicast of L16 (type 10): Asterisk and wireshark disagree | (Reported by Tzafrir Cohen) | |
| [patch]external_media_address and external_signaling_address don't always honor localnet | (Reported by Walter Doekes) | |
| CDR: CDR(start,u) function won't work in cdr_custom config | (Reported by Jacek Konieczny) | |
| res_smdi: convert to astobj2 | (Reported by Corey Farrell) | |
| chan_sip: Asterisk crashing when subscription doesn't get set | (Reported by Bryan Walters) | |
| SDP origin attribute modified when issuing re-INVITE because of directmedia=yes | (Reported by saghul) | |
| alembic: prune_on_boot fix erroneous | (Reported by Florian Floimair) | |
| When in queue on g722 with interruptions, music on hold can get stuck and no longer play | (Reported by Jens T.) | |
| nat/external_media settings ignored in 14.4.1 | (Reported by Christopher van de Sande) | |
| PJSIP external_media_address ignored if no local_net options are provided | (Reported by Matt Jordan) | |
| Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive | (Reported by Ross Beer) | |
| Crash when freeing dtls_cfg->cafile | (Reported by Richard Kenner) | |
| ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c | (Reported by Tzafrir Cohen) | |
| libc segfault upon entry into app_directory | (Reported by David Moore) | |
| Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash | (Reported by Ross Beer) | |
| core: ast_safe_system command injection possible. | (Reported by Corey Farrell) | |
| res_rtp_asterisk: Media can be hijacked even with strict RTP enabled | (Reported by Joshua C. Colp) | |
| res_rtp_asterisk: Allow remote SSRC to change due to renegotiation | (Reported by Joshua C. Colp) | |
| Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name | (Reported by James Terhune) | |
| core: Don't queue up multiple video update frames. | (Reported by Joshua C. Colp) | |
| app_minivm fails to clean up mkstemp files | (Reported by Walter Doekes) | |
| several filename bugs in Record() application | (Reported by klaus3000) | |
| alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table | (Reported by Florian Floimair) | |
| Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used | (Reported by Torrey Searle) | |
| When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used. | (Reported by Jim Van Meggelen) | |
| When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored | (Reported by Eelco Brolman) | |
| bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU | (Reported by Richard Mudgett) | |
| [patch] app_queue: Wrong queue stat calculation | (Reported by sungtae kim) | |
| XMPP OAuth not working due to inverted logic | (Reported by Michael Kuron) | |
| res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical | (Reported by Mark Thompson) | |
| If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed | (Reported by Seán C. McCord) | |
| manager: hook event is not being raised | (Reported by Kevin Harwell) | |
| Either asterisk or pjproject isn't re-using tcp connections (again) | (Reported by George Joseph) | |
| IPv6 receive address in message doesn't include brackets | (Reported by Scott Griepentrog) | |
| [patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used | (Reported by Torrey Searle) | |
| Asymmetric codecs when asymmetric_rtp_codec=no | (Reported by Jesse Ross) | |
| Make --with-pjproject-bundled the default for Asterisk 15 | (Reported by George Joseph) | |
| RTP session is not fully destroyed on channel hangup | (Reported by Matt Jordan) | |
| bridge: Crash when mapping streams | (Reported by Joshua C. Colp) | |
| channel: requester leaks joint_cap on success. | (Reported by Corey Farrell) | |
| res_pjsip_session: Handling of 'msid' is incorrect | (Reported by Kevin Harwell) | |
| res_pjsip: parse/add msid attribute when webrtc is enabled | (Reported by Kevin Harwell) | |
| Asterisk 15.0.0-Beta1 does not compile | (Reported by Ira Emus) | |
| res_pjsip: PJSIP presence - missing braces around the status element in XML | (Reported by Abraham Liebsch) | |
| Asterisk won't compile on Fedora 26 with devmode enabled. | (Reported by Corey Farrell) | |
| res_pjsip: TLS connection not stable | (Reported by Ian Gilmour) | |
| Applications ARI: Unsubscribe action for deviceStates does not remove old subscriptions properly | (Reported by Sergej Kasumovic) | |
| say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds. | (Reported by Nicolas Riendeau) | |
| sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5 | (Reported by Corey Farrell) | |
| bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues. | (Reported by Joshua C. Colp) |
Improvements made in this release:
-----------------------------------
| [patch] Add IPv6 Support for DUNDi | (Reported by Adam Secombe) | |
| Dialplan Function for Checking Parking Lot Slot | (Reported by JoshE) | |
| [PATCH] Add predial handler to app_queue | (Reported by Kristian Høgh) | |
| [patch] BuildSystem: Enable autotools in Solaris 11. | (Reported by Alexander Traud) | |
| Ten seconds of silence after mp3 playback | (Reported by Sam Wierema) | |
| [patch] res_rtp_asterisk: Allow OpenSSL configured with no-deprecated. | (Reported by Alexander Traud) | |
| [patch] res_crypto: Allow OpenSSL configured with no-deprecated. | (Reported by Alexander Traud) | |
| app_confbridge: Add talking indicator for ConfBridgeList AMI response | (Reported by William McCall) | |
| documentation: Error on wiki description of Asterisk 13 "MeetmeMute" event | (Reported by Alessandro Polidori) | |
| ast_coredumper: Fix OUTPUT directory | (Reported by Ted G) | |
| [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with no-deprecated. | (Reported by Alexander Traud) | |
| res_hep: Allow create_address to resolve a provided hostname | (Reported by Sebastian Gutierrez) | |
| [patch] Add DragonFly BSD. | (Reported by Alexander Traud) | |
| wrong automatic ras address assignment if multihomed | (Reported by Dmitry Melekhov) | |
| cppcheck identifies redundant "if" | (Reported by Ilya Shipitsin) | |
| Enable in-dialog NOTIFY on chan_pjsip channels | (Reported by Nathan Bruning) | |
| [patch] install_prereq: Add Slackware (somehow). | (Reported by Alexander Traud) | |
| [patch] install_prereq: Add Gentoo Linux. | (Reported by Alexander Traud) | |
| [patch] install_prereq: Add Arch Linux. | (Reported by Alexander Traud) | |
| [patch] install_prereq: Add SUSE. | (Reported by Alexander Traud) | |
| [patch] libsrtp-2.1.x support | (Reported by Alexander Traud) | |
| [patch] BuildSystem: Add NetBSD. | (Reported by Alexander Traud) | |
| PJSIP: Update bundled PJPROJECT to version 2.7.2 | (Reported by Richard Mudgett) | |
| [patch] install_prereq: Add NetBSD. | (Reported by Alexander Traud) | |
| [patch] BuildSystem: Allow newer autotools on OpenBSD. | (Reported by Alexander Traud) | |
| [patch]contrib/scripts: add a way to migrate from chan_sip to chan_pjsip realtime | (Reported by Torrey Searle) | |
| Add new AMI Event for Load, Unload | (Reported by sungtae kim) | |
| app_confbridge: Add Muted to ConfbridgeJoin and channel snapshot headers to ConfbridgeList AMI events | (Reported by Richard Mudgett) | |
| app_confbridge/bridge_softmix: When channel muted report talking stopped if was talking. | (Reported by Richard Mudgett) | |
| Reduce verbosity while loading PBX extensions. | (Reported by Ludovic Gasc (Eyepea)) | |
| [patch] Add config option to play a prompt to the "winner" in app_followme | (Reported by Graham Mainwaring) | |
| res_pjsip: Add new AMI Action for PJSIPShowAors | (Reported by sungtae kim) | |
| Allow wrapuptime to be set for each queue member | (Reported by Rodrigo Ramirez Norambuena) | |
| cdr.c: Minor code optimizations. | (Reported by Richard Mudgett) | |
| Add new object for VoicemailUserEntry | (Reported by sungtae kim) | |
| 3PCC patch for AMI "SIPnotify" | (Reported by Yasuhiko Kamata) | |
| [PATCH] When failing to acquire target during attended transfer, display wanted extension | (Reported by Niklas Larsson) | |
| app_voicemail: Add new object for VoicemailUserEntry | (Reported by sungtae kim) | |
| ast_coredumper: allow pointing out the asterisk binary explicitly | (Reported by Tzafrir Cohen) | |
| Compilation warning for invert.c (array subscript is above array bounds) | (Reported by Marcello Ceschia) | |
| pjproject bundled: Don't disable assertions when --enable-dev-mode is used. | (Reported by Corey Farrell) | |
| Upgrade bundled PJPROJECT to 2.7 | (Reported by Richard Mudgett) | |
| CDR performance needs improvement. | (Reported by Richard Mudgett) | |
| [patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE | (Reported by David J. Pryke) | |
| alembic: Add support for Microsoft SQL server | (Reported by Florian Floimair) | |
| Enable CHANNEL function to get from and to tag from SIP Headers | (Reported by Andre Nazario) | |
| Google OAuth 2.0 support for XMPP / Motif | (Reported by Andrey) | |
| Support for GMIME 3.0 | (Reported by Tzafrir Cohen) | |
| [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip | (Reported by Torrey Searle) |
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
Thank you for your continued support of Asterisk!
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