The Asterisk Development Team would like to announce the first release candidate of Asterisk 18.0.0. This release candidate is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 18.0.0-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: ----------------------------------- * ASTERISK-28589 - chan_sip: Depending on configuration an INVITE can alter Addr of a peer (Reported by Andrey V. T.) * ASTERISK-28580 - Bypass SYSTEM write permission in manager action allows system commands execution (Reported by Eliel Sardañons) * ASTERISK-28495 - res_pjsip_t38: 200 OK with SDP answer with declined stream causes crash (Reported by Alexei Gradinari) New Features made in this release: ----------------------------------- * ASTERISK-6863 - [patch] allow Asterisk to set high ToS bits as non-root on Linux (Reported by Matt Addison) * ASTERISK-17491 - CURLOPT() needs a "followlocation" parameter / "maxredirs" doesn't do anything (Reported by candrews) * ASTERISK-28639 - res_pjsip_endpoint_identifier_ip: Add ability to match on source port (Reported by Sean Bright) * ASTERISK-28614 - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead of only "sending" (Reported by lvl) * ASTERISK-28613 - func_curl: CURLOPT cannot set Content-Type header (Reported by Martin Tomec) * ASTERISK-28533 - func_jitterbuffer: Add support for video synchronization (Reported by Joshua C. Colp) * ASTERISK-17808 - [patch] Unregister a realtime moh class (Reported by Byron Clark) * ASTERISK-28489 - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From header URI domain (Reported by Stas Kobzar) Bugs fixed in this release: ----------------------------------- * ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY events (Reported by Ove Aursand) * ASTERISK-29043 - app_queue: Leave empty sometimes not recorded as abandoned (Reported by Kfir Itzhak) * ASTERISK-29042 - res_parking: Parker UUID is no longer copied (Reported by Misha Vodsedalek) * ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported by Joseph Ades) * ASTERISK-29046 - pbx: Deadlock when doing a reload, while simultaneously doing an ExtensionState on a pattern match hint that ends up adding an extension (Reported by Ramarajan) * ASTERISK-29040 - res_speech: Assertion on format (Reported by Nickolay V. Shmyrev) * ASTERISK-29001 - chan_pjsip does not process or forward 181 responses (Reported by Torrey Searle) * ASTERISK-29034 - Lastpause of realtime members is reseting (Reported by Evandro César Arruda) * ASTERISK-27273 - app_voicemail: When a voicemail is marked as "Urgent", it is not sent by email/processed by the mailcmd command (Reported by Leandro Dardini) * ASTERISK-29033 - res_pjsip_session: Aggressively terminates session on failed re-INVITE (Reported by Joshua C. Colp) * ASTERISK-28974 - res_rtp_asterisk: T.140 messages have appended RTP string to each message block. (Reported by Thomas Johnson) * ASTERISK-29011 - chan_sip: ToHost property not cleared on reload (Reported by Dennis) * ASTERISK-29021 - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified versions (Reported by cmaj) * ASTERISK-28927 - Asterisk crash in music on hold (Reported by David Cunningham) * ASTERISK-28973 - Malformed IP address in SDP of 2nd SIP timer triggered INVITE when NAT is active (UDP transport with external_media_address) (Reported by Michael Neuhauser) * ASTERISK-28995 - res_pjsip_registrar: Expires on statically configured contacts is not correct (Reported by tootai) * ASTERISK-28987 - BridgeCreated ARI event shows wrong video_mode info (Reported by sungtae kim) * ASTERISK-28978 - acl: named_acl rule misconfiguration results in segfault on reading rule from realtime (Reported by Andrew Yager) * ASTERISK-28975 - res_http_websocket: Text payload data doesn't necessary include trailing zero (Reported by Nickolay V. Shmyrev) * ASTERISK-28951 - Inconsistent behaviour queues.conf when there is (not) a [general] section (Reported by Walter Doekes) * ASTERISK-28965 - res_pjsip: Apply outbound proxy to static contacts on AOR (Reported by Joshua C. Colp) * ASTERISK-28930 - ./configure --without-ssl build failure (Reported by Jaco Kroon) * ASTERISK-28957 - chan_sip: chan_sip does not process 400 response to an INVITE. (Reported by Frederic LE FOLL) * ASTERISK-28886 - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2 (Reported by Jared Smith) * ASTERISK-28888 - res_corosync: causes asterisk crash in huge distributed environment. (Reported by Università di Bologna - CESIA VoIP) * ASTERISK-28954 - StreamEcho() only returns 1 active stream (Reported by Bill Kervaski) * ASTERISK-28955 - "setvar" doesn't work properly in dahdi-channels.conf (Reported by Marin Odrljin) * ASTERISK-28953 - res_pjsip_session: Preserve stream label (Reported by Joshua C. Colp) * ASTERISK-28942 - res_sorcery_memory_cache: Individual object expiration behaves unexpectedly with full backend caching (Reported by Joshua C. Colp) * ASTERISK-28950 - Stale code in app_queue to check untouched channel (Reported by Walter Doekes) * ASTERISK-28644 - Stale comment in app_queue about ring_entry exception (Reported by Walter Doekes) * ASTERISK-28952 - Queue wrapuptime sometimes not respected (based on stale lastcall time) (Reported by Walter Doekes) * ASTERISK-28938 - core_unreal / core_local: Add support for multistream and re-negotiation (Reported by Joshua C. Colp) * ASTERISK-28948 - ARI channel create doesn't referencing the channel_id parameter (Reported by sungtae kim) * ASTERISK-28939 - res_rtp_asterisk: Don't have send/receive buffers on non-WebRTC (Reported by Joshua C. Colp) * ASTERISK-28944 - bridge_softmix: Transitioning a stream from inactive -> sendrecv/sendonly doesn't re-negotiation (Reported by Joshua C. Colp) * ASTERISK-28923 - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury Kirsanov) * ASTERISK-28940 - /channels/create doesn't get any parameters from the body (Reported by sungtae kim) * ASTERISK-28936 - res_pjsip: crash when dialing non-sip uri (Reported by Walter Doekes) * ASTERISK-28900 - res_fax: Double frame free when gateway in use with off-nominal format usage (Reported by Gregory Massel) * ASTERISK-28929 - pjproject_bundled: Honor --without-pjproject. (Reported by Alexander Traud) * ASTERISK-28932 - res_pjsip_logger writing too big packets (Reported by nappsoft) * ASTERISK-28920 - bridge show all causes crash (Reported by sungtae kim) * ASTERISK-28921 - Wrong return value check for fwrite when writing to pcap file (Reported by nappsoft) * ASTERISK-28794 - res_pjsip: Crash when escaping during URI printing (Reported by nappsoft) * ASTERISK-28884 - x-ast-orig-host not filtered out from request URI and To header (Reported by nappsoft) * ASTERISK-28871 - res_pjsip_session: Unnecessary re-Invite on call answer (Reported by Alexei Gradinari) * ASTERISK-28903 - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES. (Reported by Alexander Traud) * ASTERISK-28898 - bridge_softmix: Conference bridge not passing silent rtp packets (Reported by Jonathan Hunter) * ASTERISK-28892 - res_musiconhold: Module res_musiconhold throws false warning (Reported by Nicholas John Koch) * ASTERISK-28904 - RTP ICE leaks the memory (Reported by sungtae kim) * ASTERISK-26780 - res_pjsip: PJSIP Registration Fails when transport=transport-udp6 (Reported by Peter Sokolov) * ASTERISK-28854 - SIGSEGV when pjsip show history encounters IPV6 address (Reported by Roger James) * ASTERISK-28797 - [patch] tcptls: Fix notice when TLS is enabled but not configured. (Reported by Alexander Traud) * ASTERISK-28804 - [patch] app_osplookup.c: Avoid a format truncation. (Reported by Alexander Traud) * ASTERISK-28776 - Non async-signal-safe syscalls used after fork before exec (Reported by nappsoft) * ASTERISK-28870 - streams: One memory leak and one issue cloning streams (Reported by George Joseph) * ASTERISK-28829 - app_queue: leaking stasis subscription when Redirecting call (Reported by lvl) * ASTERISK-25844 - app_queue: Ghost channels in "core show channels" output (Reported by Etienne Lessard) * ASTERISK-28859 - pjsip: Increase maximum candidate count (Reported by Joshua C. Colp) * ASTERISK-22920 - Crash while Forwarding from TLS extension with CHANNEL args secure_bridge_media and secure_bridge_signaling (Reported by Shlomi Gutman) * ASTERISK-28852 - Unprotected access to nochecksums variable, causes build failures (Reported by Guido Falsi) * ASTERISK-28848 - app_fax: Compile. (Reported by Alexander Traud) * ASTERISK-28846 - stream: Enforce formats immutability (Reported by Joshua C. Colp) * ASTERISK-28847 - ARI channels cuts the endpoint string over 80 characters (Reported by sungtae kim) * ASTERISK-28811 - Crash occurs when fax session switches from T.38 to audio (Reported by Alexey Vasilyev) * ASTERISK-28839 - Sporadic crashes with Segmentation fault (Reported by Joeran Vinzens) * ASTERISK-28835 - IPv6 addresses in SDP incorrectly formatted (Reported by Daniel Heckl) * ASTERISK-28372 - Asterisk REPLY Wrong Contact header port (TCP) (Reported by Anton Satskiy) * ASTERISK-24428 - Document that Asterisk will use the default SIP ports (5060 for TCP, 5061 for TLS) if the extern option variants aren't used (Reported by sstream) * ASTERISK-28838 - AST_MODULE_INFO requires, MODULEINFO does not mention (Reported by Alexander Traud) * ASTERISK-28841 - app_confbridge: Add support for disabling text messaging for a user (Reported by Joshua C. Colp) * ASTERISK-28837 - pjproject_bundled: Honor --without-pjproject. (Reported by Alexander Traud) * ASTERISK-28827 - res_rtp_asterisk: Loop when receive buffer is flushed by a received packet that is also in receive buffer with NACK (Reported by nappsoft) * ASTERISK-27195 - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and TLS sockets (Reported by Joshua Roys) * ASTERISK-28826 - res_rtp_asterisk: Duplicate seqnos being added to send buffer with NACK (Reported by nappsoft) * ASTERISK-28812 - First DTMF is not get (Reported by Bernard Merindol) * ASTERISK-28758 - pjsip startup errors when using "with-ssl" configure option (Reported by Patrick Wakano) * ASTERISK-28824 - BuildSystem: Search for Python/C API when possibly needed only. (Reported by Alexander Traud) * ASTERISK-27717 - [patch] BuildSystem: In NetBSD, the Python Programming Language is python-2.7. (Reported by Alexander Traud) * ASTERISK-28817 - chan_pjsip: constant DTMF tone if RTP is not setup yet (Reported by Kevin Harwell) * ASTERISK-28819 - [patch] bridge_softmix_binaural: Show state in menuselect. (Reported by Alexander Traud) * ASTERISK-28816 - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers. (Reported by Alexander Traud) * ASTERISK-28818 - [patch] BuildSystem: Allow space in path. (Reported by Alexander Traud) * ASTERISK-28809 - [patch] res_rtp_asterisk: Avoid absolute value on unsigned subtraction. (Reported by Alexander Traud) * ASTERISK-28796 - func_channel: cannot read fields exten, context, userfield, channame from dialplan (Reported by Sébastien Duthil) * ASTERISK-28803 - [patch] chan_unistim: Avoid tautological warnings with clang. (Reported by Alexander Traud) * ASTERISK-28808 - [patch] test_stasis: Avoid always true warning with clang. (Reported by Alexander Traud) * ASTERISK-28056 - res_pjsip: Incorrect endpoint status after endpoint synchronization for a specific AOR (Reported by Jason Hord) * ASTERISK-28795 - channel: write to a stream on multi-frame writes (Reported by Kevin Harwell) * ASTERISK-28789 - test_utils: incorrectly printing error 'declined to load' (Reported by Alexander Traud) * ASTERISK-28788 - func_aes: incorrectly printing error 'declined to load' (Reported by Alexander Traud) * ASTERISK-28790 - Crash during conference call using confbridge and video (Reported by Pascal Cadotte Michaud) * ASTERISK-16676 - DAHDIRAS fails to properly initiate pppd unless asterisk is running as root (Reported by Jaco Kroon) * ASTERISK-21205 - [patch] dundi_read_result crash due to negative number (Reported by Jaco Kroon) * ASTERISK-28784 - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream (Reported by Joshua C. Colp) * ASTERISK-28743 - Asterisk is crashing if the 200 OK with SDP (Reported by sungtae kim) * ASTERISK-28783 - res_pjsip_session: Allow default non-audio streams to have reflected state (Reported by Joshua C. Colp) * ASTERISK-28774 - chan_pjsip's rtptimeout is erroneously triggered during direct-media (native_rtp) bridge (Reported by Michael Neuhauser) * ASTERISK-20325 - Comments in configs/func_odbc.conf.sample are not consistent with examples. Missing examples. (Reported by Olivier Krief) * ASTERISK-28780 - app_mixmonitor: Memory leak due to race condition between AMI MixMonitor and hangup (Reported by Joshua C. Colp) * ASTERISK-28773 - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active (Reported by Torrey Searle) * ASTERISK-28769 - DTLS Handshake Fails to Occur if ice_support is enabled but not used (Reported by Torrey Searle) * ASTERISK-28759 - A non negotiated rtp frame causes call disconnection when there is a SSRC change (Reported by Paulo Vicentini) * ASTERISK-26711 - func_enum: ENUM code wrong case (Reported by Vitold) * ASTERISK-23407 - Fix the FSF address in the headers of lots of pjproject files (Reported by Jared Smith) * ASTERISK-19460 - [patch] Function TXTCIDNAME never actually makes DNS calls and always returns an empty string (Reported by George Joseph) * ASTERISK-28766 - PJSIP blind transfer not completed after using Proceeding() (Reported by lvl) * ASTERISK-28764 - res_rtp_asterisk: Improve NACK support and seqno handling (Reported by Joshua C. Colp) * ASTERISK-28755 - SIP/Stasis: SIP headers not transmitted in the "variables" field (Reported by Jean Aunis - Prescom) * ASTERISK-28685 - check_expr2: linking (when hardening) and cross-compiling troubles (Reported by Sebastian Kemper) * ASTERISK-28754 - ASTERISK-28738 Causes Audio Issue After Hold (Reported by Ross Beer) * ASTERISK-28697 - res_pjsip: Named ACL does not update on reload if changed (Reported by Timothy Vanderaerden) * ASTERISK-28746 - res_pjsip_outbound_registration keeps retrying the first entry in a SRV record set (Reported by George Joseph) * ASTERISK-28716 - ICE: pjnath shouldn't wait for ICE to complete before allowing sending (Reported by Benjamin Keith Ford) * ASTERISK-28738 - Incorrect state machine used when MOH_PASSTHRU is used (Reported by Torrey Searle) * ASTERISK-28742 - res_rtp_asterisk: static for audio due to incomplete dtls/srtp setup (Reported by Kevin Harwell) * ASTERISK-28735 - Realtime MoH Unknown format '' -- defaulting to SLIN (Reported by Ross Beer) * ASTERISK-28730 - res_pjsip_session: Fix out of order session refreshes (Reported by Joshua C. Colp) * ASTERISK-26955 - pjsip: SIP Packets with Via "received=" Containing IPv6 Address Delimited by "[]" Rejected (Reported by Peter Sokolov) * ASTERISK-28718 - chan_sip: Returns 403 if RTP ports are depleted, should return 503 (Reported by Walter Doekes) * ASTERISK-28713 - res_stasis_playback: Error building JSON (Reported by Sébastien Duthil) * ASTERISK-28714 - REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer) * ASTERISK-26082 - res_pjsip_messaging: MessageSend Content-Type can't be changed (Reported by Alex) * ASTERISK-28423 - ARI causes STASIS Deadlock (Reported by Ross Beer) * ASTERISK-28679 - stasis application is destroyed after its creation (Reported by Francois Blackburn) * ASTERISK-25421 - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the error when sending (Reported by Dmitriy Serov) * ASTERISK-28686 - chan_sip strictrtp=yes fails when media source is changed: no audio (Reported by Walter Doekes) * ASTERISK-28139 - RTP Stream Incorrect Payload Type Causes Asterisk To Drop Calls (Reported by Paul Brooks) * ASTERISK-28677 - CDR billsec is always 0 for transferred calls (Reported by Maciej Michno) * ASTERISK-28702 - chan_dahdi: holding a channel via flash to dialtone times out after 0:16:40 (Reported by Andrew Siplas) * ASTERISK-24484 - Update documentation for statsd module - usage requirements unclear (Reported by Dan Jenkins) * ASTERISK-28706 - silk 24hHz doesn't show up in 'core show translation' output (Reported by Sean Bright) * ASTERISK-28695 - core: minmemfree watermark uses free RAM, not available RAM (Reported by Kevin Flyn) * ASTERISK-28693 - chan_sip: SIP MESSAGE beginning with a whitespace appears empty in the dialplan (Reported by Frank Matano) * ASTERISK-23739 - [patch]Segfault forwarding voicemail with ODBC storage enabled and realtime voicemail_data is used (Reported by Stas Kobzar) * ASTERISK-27622 - empty voicemail.conf required for ARA (realtime) voicemail to leave message (Reported by Jim Van Meggelen) * ASTERISK-21794 - CLI command 'realtime update2' syntax failure when using according to usage help (Reported by Cedric BASSAGET) * ASTERISK-28349 - Pause reason not reported in QueueMember AMI event (Reported by Niksa Baldun) * ASTERISK-25429 - res_pjsip_endpoint_identifier_ip: Document support for hostnames (Reported by Joshua C. Colp) * ASTERISK-27775 - res_pjsip_notify: Multiple Event headers can be present instead of just one (Reported by AvayaXAsterisk) * ASTERISK-28682 - app_record: Lack of `beep` audio file causes application to return error and hangup (Reported by Corey Farrell) * ASTERISK-28507 - Wiki docs missing for MessageWaiting (Reported by David M. Lee) * ASTERISK-27759 - res_pjsip_pubsub: Subscription persistence does not preserve XML <dialog-info> version number (Reported by Bryan Nelson) * ASTERISK-28605 - chan_dahdi: Deadlock in Hangup Scenarios with concurrent command pri show span X (Reported by Dirk Wendland) * ASTERISK-28633 - stasis bridge topic leak (Reported by Joeran Vinzens) * ASTERISK-28492 - pjsip reload not reloading wizard endpoint/pickup_group endpoint/call_group (Reported by Jean-Denis Girard) * ASTERISK-28562 - SIP WSS message not processed until next frame arrives (Reported by Robert Sutton) * ASTERISK-28667 - Asterisk ignores parsing of config files if a Byte order mark is present (Reported by Robin Leffmann) * ASTERISK-28625 - Playback of local files impacted by large media cache (Reported by Kevin Reeves) * ASTERISK-27243 - contrib: valgrind.supp doesn't suppress what it's supposed to due to invalid syntax (Reported by Richard Kenner) * ASTERISK-28664 - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by Pascal Cadotte Michaud) * ASTERISK-28636 - app_chanisavail+cdr: ChanIsAvail sometimes fails to deactivate CDR. (Reported by Frederic LE FOLL) * ASTERISK-28604 - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0 (Reported by George Joseph) * ASTERISK-28659 - res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them (Reported by nappsoft) * ASTERISK-28660 - res_fax: wrap Asterisk initiated negotiation with config option (Reported by Kevin Harwell) * ASTERISK-28626 - Missing arguments in PJSIP_CONTACT function documentation (Reported by Pascal Cadotte Michaud) * ASTERISK-28609 - Memory Leak in res_rtp_asterisk.c (Reported by Ted G) * ASTERISK-28651 - chan_sip logs errors on tx to non-existent TCP connections (Reported by Jaco Kroon) * ASTERISK-28502 - chan_pjsip incorrectly re-writes REGISTER 200 Response Contact (Reported by Ross Beer) * ASTERISK-28641 - res_pjsip Segfaults when realtime configuration to an AOR points to a not existent AOR (Reported by Ross Beer) * ASTERISK-28647 - chan_sip: RTP frames not transmitted after emitting a COLP (Reported by Jean Aunis - Prescom) * ASTERISK-28637 - chan_sip+native_bridge_rtp: directmedia compatibility check failure when negociated ptime is not default ptime. (Reported by Frederic LE FOLL) * ASTERISK-28445 - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on hangup when TEST_FRAMEWORK enabled (Reported by Bernhard Schmidt) * ASTERISK-28631 - res_parking: Doesn't park when parkee and parker are the same (Reported by Ross Beer) * ASTERISK-28621 - Enforce T.38 error correction mode at 200 ok received (Reported by Salah Ahmed) * ASTERISK-28624 - res_pjsip_outbound_registration: add SRV failover (Reported by Kevin Harwell) * ASTERISK-28608 - app_amd: Use time calculation to calculate timeout (Reported by Michael Cargile) * ASTERISK-28615 - chan_dahdi: PRI span status may stay "Down, Active" after a short alarm (Reported by Frederic LE FOLL) * ASTERISK-28576 - res_rtp_asterisk: ICE Completion Crash when sent packet length doesn't match (Reported by Joshua Elson) * ASTERISK-26481 - FILE function grabs garbage along with read data when target line has no newline (Reported by Jonathan Harris) * ASTERISK-28618 - bridge_softmix: hold not cleared when joining a softmix bridge (Reported by Kevin Harwell) * ASTERISK-28616 - parking: Deadlock when multi call parking (Reported by Joshua C. Colp) * ASTERISK-28572 - Memory leaks in res_calendar_exchange and res_calendar_icalendar (Reported by Yoooooo Ha) * ASTERISK-28585 - ari/resource_events: Crash in event session cleanup (Reported by Kevin Harwell) * ASTERISK-28590 - utils.c throws repeated warnings; "pthread_attr_setstacksize: Invalid argument" (Reported by Speed Dial Dave) * ASTERISK-28578 - race condition on pjsip channelstats command (Reported by Salah Ahmed) * ASTERISK-28571 - cdr_pgsql: accesses obsolete (and finally removed) column (Reported by Christoph Moench-Tegeder) * ASTERISK-28575 - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson) * ASTERISK-28574 - pjproject fails to build on 16.6.0, works on 16.5 (Reported by Niklas Larsson) * ASTERISK-28561 - Asterisk Deadlocks (Reported by Aheliotech) * ASTERISK-28086 - chan_pjsip: Crash when initiating PlayDTMF over AMI (Reported by Jeremiah Gadd) * ASTERISK-28552 - res_pjsip_mwi: Frack during unload on unsolicited_mwi container (Reported by Kevin Harwell) * ASTERISK-28566 - CDR backend unload problem during active call(s) (Reported by Marian Piater) * ASTERISK-28553 - stasis.c: Crash during unload (Reported by Kevin Harwell) * ASTERISK-28544 - Wrong contact representation in ipv6 mode (Reported by Jørgen H) * ASTERISK-28534 - Segmentation fault when there is no priority for an extension (Reported by Timothy Vanderaerden) * ASTERISK-28463 - res_pjsip_path: Crash when invalid contact is configured (Reported by Juan Martin) * ASTERISK-28521 - pjsip: Memory Leak (Reported by Mark) * ASTERISK-28523 - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière) * ASTERISK-28536 - Asterisk release candidates fail to build on FreeBSD (Reported by Guido Falsi) * ASTERISK-28538 - chan_pjsip: Deadlock on fax detection (Reported by Joshua C. Colp) * ASTERISK-28497 - func_odbc: truncating Unicode string on readsql (Reported by Boris P. Korzun) * ASTERISK-23756 - setvar directive when used in template and a child of said template, results in duplicate variable names (Reported by Michael Goryainov) * ASTERISK-28527 - ChanIsAvail() creates a CDR if unanswered=yes is set in cdr.conf (Reported by Frederic LE FOLL) * ASTERISK-28525 - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up (Reported by Frederic LE FOLL) * ASTERISK-28511 - codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32 (Reported by Ruddy G) * ASTERISK-28499 - translate: Crash when frame does not have a "src" field set (Reported by Gregory Massel) * ASTERISK-25592 - chan_unistim: Clang Warning: variable sized type not at end of a struct (Reported by Alexander Traud) * ASTERISK-28488 - pjsip mwi: n+1 sip notify's sent on re-register (Reported by Chris Savinovich) * ASTERISK-28509 - PJSIP cnonce generated on Linux contains 36 characters, NEC only supports up to 32 characters (Reported by Dan Cropp) * ASTERISK-28505 - app_voicemail/IMAP: segfault in leave_voicemail because not checking mailstream (Reported by Alexei Gradinari) * ASTERISK-28487 - compile menuselect on gentoo (Reported by Kilburn) * ASTERISK-28472 - Asterisk occasionally passes a NULL as srtp->session to srtp_protect/unprotect causing SEGV (Reported by Jonas Swiatek) * ASTERISK-28498 - cel / cdr: Event times may be incorrect (Reported by Joshua C. Colp) * ASTERISK-28480 - json integer overflow in ssrc and timestamp (Reported by Salah Ahmed) * ASTERISK-28228 - res_pjsip: pjsip show contacts prints double entries (Reported by Ian Jones) * ASTERISK-28483 - packet lost on UDPTL wrap around (Reported by Torrey Searle) Improvements made in this release: ----------------------------------- * ASTERISK-28959 - res_pjsip: Added option for disable rport parameter set (Reported by sungtae kim) * ASTERISK-28958 - Continue reading string when ping received by websocket (Reported by Nickolay V. Shmyrev) * ASTERISK-28945 - AMI SendText - add Content-Type parameter (Reported by Kevin Harwell) * ASTERISK-28949 - res_http_websocket: Add masking to websocket client (Reported by Moises Silva) * ASTERISK-28899 - Upgrade Asterisk to bundled pjproject 2.10 (Reported by Kevin Harwell) * ASTERISK-28895 - res_pjsip_logger: Add tons'o'functionality (Reported by Joshua C. Colp) * ASTERISK-28896 - ari: Add support for specifying variables on channel create (Reported by Joshua C. Colp) * ASTERISK-28879 - pjproject has race conditions in it's build system (Reported by Guido Falsi) * ASTERISK-28866 - third-party/pjproject/configure.m4 contains bashisms (Reported by Guido Falsi) * ASTERISK-28853 - Missing include on FreeBSD (Reported by Guido Falsi) * ASTERISK-28832 - chan_mobile creates PCMA streams that make some VoIP clients crash or not render received audio (Reported by Peter Turczak) * ASTERISK-28813 - func_volume: Allow decimal numbers as parameter to improve granularity (Reported by Jean Aunis - Prescom) * ASTERISK-28777 - Codec Negotiation: add outgoing_call_offer_prefs option (Reported by Kevin Harwell) * ASTERISK-27946 - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't (Reported by Joshua Elson) * ASTERISK-28782 - Add support for Content-Disposition header in multi-part INVITES (Reported by Torrey Searle) * ASTERISK-28787 - res_pjsip_session: Decide more intelligently when to add video (Reported by Joshua C. Colp) * ASTERISK-28756 - Codec Negotiation: add incoming_call_offer_pref option (Reported by Kevin Harwell) * ASTERISK-28750 - TLS/SSL Key too small error (Reported by Martin Zeh) * ASTERISK-28733 - stream: Add support for adding/removing streams during SFU/calls (Reported by Joshua C. Colp) * ASTERISK-24798 - Documentation - Clarify That Format Is Set By File Name Extension In MixMonitor (Reported by xrobau) * ASTERISK-28726 - install_prereq script uses the interactive mode when installing aptitude (Reported by Sylvain Afchain) * ASTERISK-28710 - Should be able to disable the /httpstatus URI in the built-in HTTP server (Reported by Sean Bright) * ASTERISK-28484 - Add AudioSocket support (Reported by Seán C. McCord) * ASTERISK-28638 - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported by cmaj) * ASTERISK-28673 - GET FULL VARIABLE documentation clarification (Reported by Jonathan Harris) * ASTERISK-28629 - [patch] Add an "inhibitCOLP" flag to the bridges REST API (Reported by Jean Aunis - Prescom) * ASTERISK-28658 - app_confbridge: Add support for setting maximum sample rate (Reported by Joshua C. Colp) * ASTERISK-28602 - res_pjsip_outbound_registration: Maximum retries reached (Reported by Daniel) * ASTERISK-28586 - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam Banks) * ASTERISK-22192 - [patch] Allow voicemail forwards with ODBC backend when format differs from attachfmt column (Reported by cmaj) * ASTERISK-28567 - Problem with ASTERISK-20207: Asterisk should clear out any .lock files in the voice mail directory on startup. (Reported by Michael) * ASTERISK-28542 - [patch] add the ability for asterisk to generate on-hold re-invites (Reported by Torrey Searle) * ASTERISK-28512 - Add pass-through support for H.265 (HEVC) codec (Reported by Florian Floimair) For a full list of changes in this release candidate, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0-rc1 Thank you for your continued support of Asterisk!
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev