The Asterisk Development Team would like to announce the first release candidate of Asterisk 16.0.0.
This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.0.0-rc1 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 candidate:
Security bugs fixed in this release:
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iostreams: Potential DoS when client connection closed prematurely (Reported by Sean Bright) |
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Username bruteforce is possible when using ACL with PJSIP (Reported by John) |
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WebSocket frames with 0 sized payload causes DoS (Reported by Sean Bright) |
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Segmentation fault occurs in asterisk with an invalid SDP fmtp attribute (Reported by Sandro Gauci) |
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Segmentation fault occurs in Asterisk with an invalid SDP media format description (Reported by Sandro Gauci) |
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Crash occurs when sending a repeated number of INVITE messages over TCP or TLS transport (Reported by Sandro Gauci) |
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SUBSCRIBE message with a large Accept value causes stack corruption (Reported by Sandro Gauci) |
New Features made in this release:
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Add the ability to read the media file type from HTTP header for playback (Reported by Gaurav Khurana) |
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Add cache_pools debug option to pjproject.conf (Reported by Richard Mudgett) |
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Add new AMI Action for PJSIPShowContacts (Reported by sungtae kim) |
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res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported by sungtae kim) |
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core: Add support for timelen parsing to ast_parse_arg and ACO. (Reported by Corey Farrell) |
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PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming INVITE Request-URI. (Reported by Richard Mudgett) |
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Add cache_media_frames debugging option. (Reported by Richard Mudgett) |
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res_pjsip: No mechanism exists to limit endpoint identification to IP only (Reported by Ben Merrills) |
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AMI : Add CancelAtxfer Action (Reported by Thomas Sevestre) |
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[New Feature] Add mute and DTMF passthrough to ARI add channel to bridge (Reported by Darren Sessions) |
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chan_sip: Access incoming SIP REFER headers in the dialplan (Reported by Kirill Katsnelson) |
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chan_sip: Dialplan function SIP_HEADERS() to complement SIP_HEADER(). (Reported by Kirill Katsnelson) |
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Add support for systemd socket activation (Reported by Corey Farrell) |
Bugs fixed in this release:
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res_pjsip: Change default transport keepalive to preserve behavior (Reported by Joshua Colp) |
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pjproject_bundled: Repair ./configure –with-ssl=PATH. (Reported by Alexander Traud) |
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BASIC-RETRANS: Implement receive (Reported by Benjamin Keith Ford) |
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res_sorcery_config: Allow object name based matching (Reported by Joshua Colp) |
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module: Remove old modules, update support levels (Reported by Joshua Colp) |
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stasis: Improve message type “Use of before init/after destruction” error (Reported by Joshua Colp) |
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srtp: rejecting short sdes lifetimes incompatible with obihai ATAs (Reported by Nick French) |
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res_pjsip: Spurious ERROR logging when printing headers in sip_msg (Reported by Nick French) |
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pjsip modules always get -O2 even when DONT_OPTIMIZE is set (Reported by George Joseph) |
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pjproject_bundled: Disable TCP/TLS keep-alives. (Reported by Alexander Traud) |
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PJSIP proposes ICE candidates on answer even if not in offer (Reported by Torrey Searle) |
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Compile fails with `IPTOS_MINCOST’ undeclared. (Reported by Alexander Traud) |
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res_pjsip_session: sdp group:BUNDLE attribute truncated (Reported by Kevin Harwell) |
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res_pjsip_pubsub: segfault in function publish_expire (Reported by Alexei Gradinari) |
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res_pjsip_rfc3326: A lot of endpoints do not correctly handle two Reason headers (Reported by Ross Beer) |
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res_pjsip_session: Initial INVITE with audio+fax results in 488 instead of declining stream (Reported by Thiago Coutinho) |
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res_pjsip_t38: ATA fails with hangupcause 58(Bearer capability not available) (Reported by Jared Hull) |
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res_pjsip_t38: Slow T.38 re-invite rejection if remote leg has T.38 disabled (Reported by Torrey Searle) |
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res_pjsip: Lock inversion in transport management (Reported by Ross Beer) |
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bridge_softmix_binaural: Enable FFTW3 in Solaris 11. (Reported by Alexander Traud) |
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res_pjsip_pubsub: apparent crash on shutdown (Reported by Kevin Harwell) |
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app_confbridge: Conference bridge and announcer channels are not removed if conference is ended as soon as it starts (Reported by Robert Mordec) |
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cdr: Deadlock with submit_scheduled_batch and submit_unscheduled_batch (Reported by Denis Lebedev) |
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pbx_dundi: Asterisk crashes when unloading module pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman) |
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AMI: Action SendText needs to use the correct thread. (Reported by Richard Mudgett) |
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res_pjsip_messaging doesn’t accept application/* content-types. (Reported by George Joseph) |
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res_pjsip_session doesn’t update media when a 200 comes in with a different port than a 183 (Reported by George Joseph) |
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uuid: Enable UUID in Solaris 11. (Reported by Alexander Traud) |
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channels: CHECK_BLOCKING is ineffective (Reported by Corey Farrell) |
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BuildSystem: Enable ./configure in Solaris 11. (Reported by Alexander Traud) |
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bootstrap.sh: find -maxdepth is not POSIX compatible. (Reported by Alexander Traud) |
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menuselect: GCC 8: restrict-qualified parameter passed and aliased. (Reported by Alexander Traud) |
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tests/test_utils: Repair ./configure –with-ssl=PATH. (Reported by Alexander Traud) |
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chan_iax2: Stops listening for traffic (Reported by Kirsty Tyerman) |
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rtp: DTMF Breaks With telephony-event/16000 (Reported by Dominic) |
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crypto.h: Repair ./configure –with-ssl=PATH. (Reported by Alexander Traud) |
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res_srtp: Repair ./configure –with-ssl=PATH. (Reported by Alexander Traud) |
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SQL fetch error on query which return 0 columns (Reported by Alexei Gradinari) |
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chan_pjsip isn’t updating hangupcause on 4XX responses (Reported by George Joseph) |
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ooh323c: GCC 8: output truncated before terminating nul. (Reported by Alexander Traud) |
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res_pjsip: Modified qualify_frequency doesn’t effect until pjsip reload (Reported by Alexei Gradinari) |
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res_fax: Deadlock when using Local channels and fax gateway (Reported by David Brillert) |
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Manager events for MeetMe have incorrectly documented key name ‘Usernum’ – should be ‘User’ (Reported by Francois Blackburn) |
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tcptls.h: Repair ./configure –with-ssl=PATH. (Reported by Alexander Traud) |
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tcptls: Allow OpenSSL configured with no-dh. (Reported by Alexander Traud) |
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tcptls: Allow OpenSSL 1.1.x configured with enable-ssl3-method no-deprecated. (Reported by Alexander Traud) |
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Codec-Change Re-INVITE during DTMF can cause marker bit error (Reported by Torrey Searle) |
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res_rtp_asterisk: Add support for abs-send-time RTP extension (Reported by Joshua Colp) |
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config/ast_destroy_realtime_fields: successful DELETE is treated as failed (Reported by Alexei Gradinari) |
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: tcptls: Repair ./configure –with-ssl=PATH. (Reported by Alexander Traud) |
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Asterisk ODBC Voicemail Prompt storage fails with recent MariaDB version. (Reported by Nic Colledge) |
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Incorrect error reported when leaving/retrieving a ODBC voicemail (Reported by Nic Colledge) |
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chan_mobile: presents incorrect inbound Caller-ID names (Reported by Brian) |
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res_pjsip_endpoint_identifier_ip: Unregister the module for headers. (Reported by Alexander Traud) |
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cli: “manager show settings” mislabels HTTP timeout as being minutes. (Reported by Corey Farrell) |
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Fix issues exposed by GCC 8 (Reported by George Joseph) |
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rtp_engine: Allow Media Formats with add_static_payload(-1) on egress again. (Reported by Alexander Traud) |
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sip_to_pjsip: Enable python3 compatibility. (Reported by Alexander Traud) |
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digest over for manager (ami) over http fails on too long uris (Reported by Jaco Kroon) |
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Macro allows an infinite loop of dialplan inclusion resulting in a crash (Reported by Tzafrir Cohen) |
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cdr_mysql creates empty records if reconnects when mysql was not up on module load (Reported by Tzafrir Cohen) |
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Asterisk got stuck while enabling “ari set debug all on” (Reported by shaurya jain) |
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chan_sip: one way / no audio with srtp (Reported by Florian Kaiser) |
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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) |
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pjsip_options: rework to make more efficient (Reported by Kevin Harwell) |
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translate: interpolated frames are not passed through (Reported by Kevin Harwell) |
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When the ooh323 debug is on there is no ringing signal to incoming calls via H323 trunk. (Reported by Dimos) |
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No “alert” or “progress” in chan_ooh323 if debug is enabled only on the module (Reported by Marco Giordani) |
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bridge_softmix / app_confbridge: Add support for combining REMB reports (Reported by Joshua Colp) |
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BuildSystem: Enable IMAP storage on FreeBSD and DragonFly BSD. (Reported by Alexander Traud) |
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app_confbridge: “core show profile bridge” does not output “sfu” when video_mode is sfu (Reported by Carlos Chavez) |
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utils/pval: Add -lBlocksRuntime for compiler clang conditionally. (Reported by Alexander Traud) |
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chan_vpb: Avoid GNU old-style field designator extension. (Reported by Alexander Traud) |
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BASIC-RETRANS: Implement send (Reported by Benjamin Keith Ford) |
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res_musiconhold: Music on hold restarts after every announcement (Reported by lvl) |
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cdr_mysql: Missing MYSQL_PORT definition (Reported by Evandro César Arruda) |
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res_pjsip_session: SDP origin does not use resolved address (Reported by John M.) |
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res_rtp_asterisk: Add support for sending RTCP feedback messages (Reported by Joshua Colp) |
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chan_sip: New Channel creation from new SIP dialog with Replaces failed to be properly tracked and destroyed (Reported by Shannon Price) |
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app_confbridge: Add ability to enable and configure REMB support (Reported by Joshua Colp) |
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PJSIP: Deadlock shutting down subscription TCP connection and sending subscription message. (Reported by Ross Beer) |
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res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported by Ross Beer) |
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res_rtp_asterisk: Add support for raising RTCP feedback messages (Reported by Joshua Colp) |
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rtp: RTCP messages with REMB trigger fast picture update (Reported by Joshua Colp) |
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Command line not being parsed correctly with getopt not from glibc (Reported by Guido Falsi) |
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configure: pjsip_evsub_set_uas_timeout not found. (Reported by Alexander Traud) |
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BuildSystem: With external editline, do not require libs for internal editline. (Reported by Alexander Traud) |
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ConfBridge: raise ConfbridgeTalking when put on hold and clear talking status (Reported by Kevin Harwell) |
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Generic PLC doesn’t work if the 2 codecs on a channel are equal (Reported by George Joseph) |
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BuildSystem: Remove unused dependency on libltdl. (Reported by Alexander Traud) |
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Make format_ogg_vorbis work on OpenBSD (Reported by Michiel van Baak) |
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BuildSystem: Enable Advanced Linux Sound Architecture (ALSA) in NetBSD. (Reported by Alexander Traud) |
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res_pjsip_rfc3326.c rfc3326_use_reason_header doesn’t account for more than one ‘Reason’ header (Reported by Ross Beer) |
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BuildSystem: Enable IMAP storage on openSUSE and Arch Linux. (Reported by Alexander Traud) |
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install_prereq: Update FreeBSD libraries. (Reported by Alexander Traud) |
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res_srtp: Add support for libsrtp2.x on openSUSE. (Reported by Alexander Traud) |
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NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt Sampson) |
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BuildSystem: Enable Better Backtraces in FreeBSD. (Reported by Alexander Traud) |
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Deprecate legacy modules (Reported by Corey Farrell) |
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uuid_generate_random detection failure (Reported by John Nemeth) |
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BuildSystem: Enable PortAudio in NetBSD. (Reported by Alexander Traud) |
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BuildSystem: AC_PATH_PROG sets to colon character when not found. (Reported by Alexander Traud) |
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res_pjsip_rfc3326: Order of ‘Reason’ headers break many endpoints (Reported by Ross Beer) |
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AMI Action VoicemailUsersList returns 0 MessageCount (Reported by Sébastien Duthil) |
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chan_sip: RTP framing issues on outgoing calls (Reported by Jean Aunis – Prescom) |
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PJSIP: Forked INVITE SDP negotiation gets one way audio. (Reported by lvl) |
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BuildSystem: Enable Lua in NetBSD. (Reported by Alexander Traud) |
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BuildSystem: Depend not implicitly but explicitly on external libraries. (Reported by Alexander Traud) |
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res_http_post: Enable GMime in NetBSD. (Reported by Alexander Traud) |
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BuildSystem: Enable autotools in NetBSD. (Reported by Alexander Traud) |
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chan_unistim: NetBSD has an incompatible struct in_pktinfo. (Reported by Alexander Traud) |
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BuildSystem: Cast any intptr_t explicitly to its proposed type. (Reported by Alexander Traud) |
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BuildSystem: Detect whether uselocale(.) is available. (Reported by Alexander Traud) |
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BuildSystem: Avoid re-defining of pthread_* on NetBSD. (Reported by Alexander Traud) |
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BuildSystem: Install init scripts on openSUSE Tumbleweed. (Reported by Alexander Traud) |
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BuildSystem: Avoid == for comparison in ./configure. (Reported by Alexander Traud) |
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app_amd.so returning TOOLONG before reaching the timeout (Reported by Michael Cargile) |
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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) |
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PJSIP: Crash during SIP attended transfer. (Reported by Bryan Walters) |
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Output from rawman truncated if output is long enough (Reported by Bojan Nemčić) |
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bridging: Sometimes cloning the stream topology causes a crash (Reported by Richard Mudgett) |
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core: If frame with unnegotiated format is read crash will occur (Reported by Sébastien Duthil) |
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Wrong remote identity and target in dialog package XML in NOTIFY (Reported by Alejandro Padilla) |
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Asterisk “doc/lang/language-criteria.txt” needs update or removal. (Reported by Rusty Newton) |
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ICE fails with no candidate nominated (Reported by Thomas Guebels) |
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rtp_engine: Load format name / mime type in uppercase again. (Reported by Alexander Traud) |
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res_pjsip: Endpoint destruction does not free DTLS configuration (Reported by Mak Dee) |
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install_prereq: Update OpenBSD libraries. (Reported by Alexander Traud) |
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res_calendar: Specialized calendars depend on symbols of general calendar. (Reported by Alexander Traud) |
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BuildSystem: Enable IMAP storage on OpenBSD. (Reported by Alexander Traud) |
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BuildSystem: Enable system provided libedit on OpenBSD. (Reported by Alexander Traud) |
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BuildSystem: Remove chan_h323 leftovers. (Reported by Alexander Traud) |
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BuildSystem: Invoke ldconfig with previous paths. (Reported by Alexander Traud) |
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BuildSystem: Do not warn when bash is not installed. (Reported by Alexander Traud) |
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chan_sip: Crash processing CANCEL request (Reported by Leandro Dardini) |
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Internal pjproject build doesn’t disable bcg729 (Reported by Stuart Henderson) |
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codecs: Add support for WebRTC iLBC 2.0. (Reported by Alexander Traud) |
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Determine if the internal editline and stdtime libraries are still relevant (Reported by George Joseph) |
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backtrace: Avoid -Wlogical-not-parentheses. (Reported by Alexander Traud) |
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install_prereq: Update Debian/Ubuntu libraries. (Reported by Alexander Traud) |
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CDR: Leaking channel snapshots allocated by stasis_channel.c (Reported by Kristijan Vrban) |
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chan_console: cannot read and write at the same time with alsa backend (Reported by Tzafrir Cohen) |
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(null) string tailing after AsyncAGIEnd AMI event (Reported by sungtae kim) |
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Null pointer Crash in PJSIP MWI (Reported by Joshua Elson) |
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res_pjsip: If SIP response is received during shutdown a crash may occur (Reported by Joshua Colp) |
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Build System: Require compiler to provide built-in support for atomic references. (Reported by Corey Farrell) |
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Subscriptions Persist After Expiration and TCP/TLS Disconnect (Reported by Ross Beer) |
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BuildSystem: Enable autotools in FreeBSD. (Reported by Alexander Traud) |
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app_voicemail: Avoid always true warnings with clang. (Reported by Alexander Traud) |
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install_prereq: Update RHEL/CentOS/Fedora libraries. (Reported by Alexander Traud) |
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core: macOS devmode build fails: variable ‘freeswap’ set but not used (Reported by David M. Lee) |
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editline: Avoid shifting a negative signed value. (Reported by Alexander Traud) |
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Problems with siren14 codec; problems with siren7 sound files. (Reported by Steve Murphy) |
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configure.ac in 1.4.37 broken with autoconf 2.60 (Reported by Stéphan Kochen) |
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install_prereq: Download latest Jansson. (Reported by Alexander Traud) |
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New module loader aborts startup if a required module declines load. (Reported by snuffy) |
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res_config_mysql: Avoid the header mysql_version.h. (Reported by Alexander Traud) |
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When running ./contrib/scripts/install_prereq install-unpackaged pjproject is installed in wrong place (Reported by PowerPBX) |
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BuildSystem: AC_CONFIG_AUX_DIR needs a directory. (Reported by Alexander Traud) |
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BuildSystem: Allow make clean all again. (Reported by Alexander Traud) |
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install_prereq: Support package manager DNF. (Reported by Alexander Traud) |
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Placing call on hold temporarily locks up set (Reported by Igor Goncharovsky) |
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BuildSystem: Use the detected name for MD5 everywhere. (Reported by Alexander Traud) |
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BuildSystem: Invoke install not in GNU but POSIX style. (Reported by Alexander Traud) |
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BuildSystem: In OpenBSD, xmlstarlet is xml. (Reported by Alexander Traud) |
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BuildSystem: Detect external library Lua in version 5.3. (Reported by Alexander Traud) |
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res_pjsip_endpoint_identifier_ip only matches against header if match by ip fails (Reported by George Joseph) |
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res_pjsip: Segfault when calling pjsip_hdr_print_on in sip_msg.c:581 (Reported by Ross Beer) |
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BuildSystem: Avoid $EUID and use id -u instead. (Reported by Alexander Traud) |
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BuildSystem: Resolve resolv.h not via Generic but Particular Header-Check. (Reported by Alexander Traud) |
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menuselect : remove obsolete TRACE_FRAMES compiler flag (Reported by Jean Aunis – Prescom) |
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res_config_pgsql: Avoid typecasting an int to unsigned char. (Reported by Alexander Traud) |
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clang 5 does not know -Wno-format-truncation (Reported by Alexander Traud) |
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app_osplookup.c: Avoid a format truncation. (Reported by Alexander Traud) |
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chan_ooh323: Avoid typecasting an int to unsigned short. (Reported by Alexander Traud) |
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chan_sip: Assumes iostream is non-NULL when it may not be (Reported by Lubos Dolezel) |
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translate: Avoid absolute value on unsigned substraction. (Reported by Alexander Traud) |
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res_pjsip_session: Improve WebRTC interop with bundling during renegotiation (Reported by Joshua Colp) |
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res_curl: Avoid error message on unload. (Reported by Alexander Traud) |
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clang 5.0: implicit conversion to char changes value to negative. (Reported by Alexander Traud) |
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bridge_softmix: Avoid warning about an uninitialized variable. (Reported by Alexander Traud) |
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editline: Avoid comparison between pointer and zero character constant. (Reported by Alexander Traud) |
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codec_gsm: Avoid shifting a negative signed value. (Reported by Alexander Traud) |
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Asterisk configure fails on ‘cannot find ptlib-config’, despite ptlib-config existing (Reported by Rusty Newton) |
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chan_ooh323: Limit outgoinglimit to positive values as intended. (Reported by Alexander Traud) |
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ooh323cDriver: Fix typo in header guard. (Reported by Alexander Traud) |
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Avoid obsolete warnings on autoconf. (Reported by Alexander Traud) |
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Modules need to ensure that any functions, apps, AMI actions, etc. they register are unregistered if the module declines loading (Reported by Mark Michelson) |
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‘cdr submit’ fails: batch mode not enabled. (Reported by Tzafrir Cohen) |
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ICE candidate parser – ICE foundation parsing too short (Reported by Michele Prà) |
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Datastore: Implement automatic module references. (Reported by Corey Farrell) |
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Asterisk Turkish Language Set Problem (Reported by Halil İbrahim YILDIZ) |
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Documentation fix – MASTER_CHANNEL Unexpected Behaviour (Reported by Shane Mitchell) |
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Compiler optimizations can break module load sequence. (Reported by abelbeck) |
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Security: Authenticated SUBSCRIBE without Contact crashes asterisk (Reported by Ross Beer) |
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Typo’s (Reported by Walter Doekes) |
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bridge: Old channel video source not set to NULL after unref (Reported by Richard Kenner) |
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DNS: Unexpected rr_type can cause crash (Reported by Corey Farrell) |
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AMI bridge of channels results in MOH not destroyed and robotic audio on one channel (Reported by Zane Conkle) |
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chan_console: ‘set active’ fails to work (Reported by Tzafrir Cohen) |
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Asterisk Hangs with Bad file descriptor on read() (Reported by Abhay Gupta) |
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ConfBridge sound_muted does not work from CLI or AMI (Reported by Thomas Frederiksen) |
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Transfer application does not work with Local channels – documentation misleading (Reported by Ivan Ullmann) |
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chan_sip: “rejected because extension not found” should be logged as a security event (Reported by Brian J. Murrell) |
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Strictrtp has issues to qualify video rtp streams (Reported by Wim De Vlaminck) |
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Coverity Report: Fix issues for error type CHAR_IO (Reported by Matt Jordan) |
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iax.conf demo peer is invalid (Reported by Tzafrir Cohen) |
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README refers to security documents that do not exist. (Reported by Corey Farrell) |
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“core set verbose” behaves strangely, can’t alias it, cli.conf example broken (Reported by Tim Ringenbach at Asteria Solutions Group) |
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crash after an invalid rtcp packet from GT48 FXS gateway (Reported by Tzafrir Cohen) |
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res_rtp_asterisk: Multiple reports in an RTCP packet will write past where it should (Reported by Vitezslav Novy) |
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Identify causes and fix pjsip/resolver/srv/failover/in_dialog/transport_tcp (Reported by Corey Farrell) |
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Queue members with hints for state_interface get stuck in “In Use” state. (Reported by Steven T. Wheeler) |
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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) |
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pjsip_options: qualify_frequency sometimes not applied on reload (Reported by John Bigelow) |
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CDR: Deadlock using AMI Originate with Variable CDR(amaflags)=… (Reported by Richard Mudgett) |
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RTP: Blind transfer direct media scenario results in one way audio. (Reported by Richard Mudgett) |
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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) |
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chan_sip: Guests disallowed via TCP (or TLS) if existing peer from same IP. (Reported by Alexander Traud) |
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pjsip: Clean up WebRTC disables (Reported by abelbeck) |
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Security: chan_skinny: Memory exhaustion if flooded with unauthenticated requests (Reported by George Joseph) |
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res_http_post: Don’t require GMIME_MAJOR_VERSION (Reported by Joshua Colp) |
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Transcoding makes bad choice in high-rate translations (Reported by Richard Kenner) |
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ARI: Updating a bridge gives wrong error message. (Reported by Frank Durden) |
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column and row headers for Signed Linear format variants in output of ‘core show translation’ are ambiguous (Reported by Rusty Newton) |
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H323 audio starts with a delay of 2 seconds. (Reported by Marco Giordani) |
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pjsip: 183 without To tag does not negotiate media (Reported by Kevin Harwell) |
|
|
ICE: server-reflexive candidates (srflx) with Dual-Stack. (Reported by Alexander Traud) |
|
|
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 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) |
|
|
tcptls: Print notice when TLS is enabled but not configured. (Reported by Alexander Traud) |
|
|
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 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) |
|
|
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 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) |
|
|
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) |
|
|
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) |
|
|
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) |
|
|
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) |
|
|
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 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) |
|
|
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 Colp) |
|
|
res_rtp_asterisk: Allow remote SSRC to change due to renegotiation (Reported by Joshua 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 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) |
|
|
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) |
|
|
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 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 Colp) |
Improvements made in this release:
———————————–
|
Dialplan Function for Checking Parking Lot Slot (Reported by JoshE) |
|
|
[PATCH] Add predial handler to app_queue (Reported by Kristian Høgh) |
|
|
BuildSystem: Enable autotools in Solaris 11. (Reported by Alexander Traud) |
|
|
Ten seconds of silence after mp3 playback (Reported by Sam Wierema) |
|
|
res_rtp_asterisk: Allow OpenSSL configured with no-deprecated. (Reported by Alexander Traud) |
|
|
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) |
|
|
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) |
|
|
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) |
|
|
install_prereq: Add Slackware (somehow). (Reported by Alexander Traud) |
|
|
install_prereq: Add Gentoo Linux. (Reported by Alexander Traud) |
|
|
install_prereq: Add Arch Linux. (Reported by Alexander Traud) |
|
|
install_prereq: Add SUSE. (Reported by Alexander Traud) |
|
|
libsrtp-2.1.x support (Reported by Alexander Traud) |
|
|
BuildSystem: Add NetBSD. (Reported by Alexander Traud) |
|
|
PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported by Richard Mudgett) |
|
|
install_prereq: Add NetBSD. (Reported by Alexander Traud) |
|
|
BuildSystem: Allow newer autotools on OpenBSD. (Reported by Alexander Traud) |
|
|
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)) |
|
|
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) |
|
|
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) |
|
|
chan_pjsip: Port SIPDtmfMode to chan_pjsip (Reported by Torrey Searle) |
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0-rc1
Thank you for your continued support of Asterisk!