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.
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:
——————————
|
chan_sip: Depending on configuration an INVITE can alter Addr of a peer (Reported by Andrey V. T.) |
|
|
Bypass SYSTEM write permission in manager action allows system commands execution (Reported by Eliel Sardañons) |
|
|
res_pjsip_t38: 200 OK with SDP answer with declined stream causes crash (Reported by Alexei Gradinari) |
New Features made in this release:
——————————
|
allow Asterisk to set high ToS bits as non-root on Linux (Reported by Matt Addison) |
|
|
CURLOPT() needs a “followlocation” parameter / “maxredirs” doesn’t do anything (Reported by candrews) |
|
|
res_pjsip_endpoint_identifier_ (Reported by Sean Bright) |
|
|
app_senddtmf: Allow “receiving” DTMF with PlayDTMF instead of only “sending” (Reported by lvl) |
|
|
func_curl: CURLOPT cannot set Content-Type header (Reported by Martin Tomec) |
|
|
func_jitterbuffer: Add support for video synchronization (Reported by Joshua C. Colp) |
|
|
Unregister a realtime moh class (Reported by Byron Clark) |
|
|
Channel variable SIPFROMDOMAIN for chan_pjsip to setup From header URI domain (Reported by Stas Kobzar) |
Bugs fixed in this release:
——————————
|
Duplicate logging in queue log for EXITEMPTY events (Reported by Ove Aursand) |
|
|
app_queue: Leave empty sometimes not recorded as abandoned (Reported by Kfir Itzhak) |
|
|
res_parking: Parker UUID is no longer copied (Reported by Misha Vodsedalek) |
|
|
chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported by Joseph Ades) |
|
|
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) |
|
|
res_speech: Assertion on format (Reported by Nickolay V. Shmyrev) |
|
|
chan_pjsip does not process or forward 181 responses (Reported by Torrey Searle) |
|
|
Lastpause of realtime members is reseting (Reported by Evandro César Arruda) |
|
|
app_voicemail: When a voicemail is marked as “Urgent”, it is not sent by email/processed by the mailcmd command (Reported by Leandro Dardini) |
|
|
res_pjsip_session: Aggressively terminates session on failed re-INVITE (Reported by Joshua C. Colp) |
|
|
res_rtp_asterisk: T.140 messages have appended RTP string to each message block. (Reported by Thomas Johnson) |
|
|
chan_sip: ToHost property not cleared on reload (Reported by Dennis) |
|
|
Fix VERSION(ASTERISK_VERSION_NUM) on certified versions (Reported by cmaj) |
|
|
Asterisk crash in music on hold (Reported by David Cunningham) |
|
|
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) |
|
|
res_pjsip_registrar: Expires on statically configured contacts is not correct (Reported by tootai) |
|
|
BridgeCreated ARI event shows wrong video_mode info (Reported by sungtae kim) |
|
|
acl: named_acl rule misconfiguration results in segfault on reading rule from realtime (Reported by Andrew Yager) |
|
|
res_http_websocket: Text payload data doesn’t necessary include trailing zero (Reported by Nickolay V. Shmyrev) |
|
|
Inconsistent behaviour queues.conf when there is (not) a [general] section (Reported by Walter Doekes) |
|
|
res_pjsip: Apply outbound proxy to static contacts on AOR (Reported by Joshua C. Colp) |
|
|
./configure –without-ssl build failure (Reported by Jaco Kroon) |
|
|
chan_sip: chan_sip does not process 400 response to an INVITE. (Reported by Frederic LE FOLL) |
|
|
chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2 (Reported by Jared Smith) |
|
|
res_corosync: causes asterisk crash in huge distributed environment. (Reported by Università di Bologna – CESIA VoIP) |
|
|
StreamEcho() only returns 1 active stream (Reported by Bill Kervaski) |
|
|
“setvar” doesn’t work properly in dahdi-channels.conf (Reported by Marin Odrljin) |
|
|
res_pjsip_session: Preserve stream label (Reported by Joshua C. Colp) |
|
|
res_sorcery_memory_cache: Individual object expiration behaves unexpectedly with full backend caching (Reported by Joshua C. Colp) |
|
|
Stale code in app_queue to check untouched channel (Reported by Walter Doekes) |
|
|
Stale comment in app_queue about ring_entry exception (Reported by Walter Doekes) |
|
|
Queue wrapuptime sometimes not respected (based on stale lastcall time) (Reported by Walter Doekes) |
|
|
core_unreal / core_local: Add support for multistream and re-negotiation (Reported by Joshua C. Colp) |
|
|
ARI channel create doesn’t referencing the channel_id parameter (Reported by sungtae kim) |
|
|
res_rtp_asterisk: Don’t have send/receive buffers on non-WebRTC (Reported by Joshua C. Colp) |
|
|
bridge_softmix: Transitioning a stream from inactive -> sendrecv/sendonly doesn’t re-negotiation (Reported by Joshua C. Colp) |
|
|
T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury Kirsanov) |
|
|
/channels/create doesn’t get any parameters from the body (Reported by sungtae kim) |
|
|
res_pjsip: crash when dialing non-sip uri (Reported by Walter Doekes) |
|
|
res_fax: Double frame free when gateway in use with off-nominal format usage (Reported by Gregory Massel) |
|
|
pjproject_bundled: Honor –without-pjproject. (Reported by Alexander Traud) |
|
|
res_pjsip_logger writing too big packets (Reported by nappsoft) |
|
|
bridge show all causes crash (Reported by sungtae kim) |
|
|
Wrong return value check for fwrite when writing to pcap file (Reported by nappsoft) |
|
|
res_pjsip: Crash when escaping during URI printing (Reported by nappsoft) |
|
|
x-ast-orig-host not filtered out from request URI and To header (Reported by nappsoft) |
|
|
res_pjsip_session: Unnecessary re-Invite on call answer (Reported by Alexei Gradinari) |
|
|
res_srtp: Answered Crypto Suite might be wrong in SDP/SDES. (Reported by Alexander Traud) |
|
|
bridge_softmix: Conference bridge not passing silent rtp packets (Reported by Jonathan Hunter) |
|
|
res_musiconhold: Module res_musiconhold throws false warning (Reported by Nicholas John Koch) |
|
|
RTP ICE leaks the memory (Reported by sungtae kim) |
|
|
res_pjsip: PJSIP Registration Fails when transport=transport-udp6 (Reported by Peter Sokolov) |
|
|
SIGSEGV when pjsip show history encounters IPV6 address (Reported by Roger James) |
|
|
tcptls: Fix notice when TLS is enabled but not configured. (Reported by Alexander Traud) |
|
|
app_osplookup.c: Avoid a format truncation. (Reported by Alexander Traud) |
|
|
Non async-signal-safe syscalls used after fork before exec (Reported by nappsoft) |
|
|
streams: One memory leak and one issue cloning streams (Reported by George Joseph) |
|
|
app_queue: leaking stasis subscription when Redirecting call (Reported by lvl) |
|
|
app_queue: Ghost channels in “core show channels” output (Reported by Etienne Lessard) |
|
|
pjsip: Increase maximum candidate count (Reported by Joshua C. Colp) |
|
|
Crash while Forwarding from TLS extension with CHANNEL args secure_bridge_media and secure_bridge_signaling (Reported by Shlomi Gutman) |
|
|
Unprotected access to nochecksums variable, causes build failures (Reported by Guido Falsi) |
|
|
app_fax: Compile. (Reported by Alexander Traud) |
|
|
stream: Enforce formats immutability (Reported by Joshua C. Colp) |
|
|
ARI channels cuts the endpoint string over 80 characters (Reported by sungtae kim) |
|
|
Crash occurs when fax session switches from T.38 to audio (Reported by Alexey Vasilyev) |
|
|
Sporadic crashes with Segmentation fault (Reported by Joeran Vinzens) |
|
|
IPv6 addresses in SDP incorrectly formatted (Reported by Daniel Heckl) |
|
|
Asterisk REPLY Wrong Contact header port (TCP) (Reported by Anton Satskiy) |
|
|
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) |
|
|
AST_MODULE_INFO requires, MODULEINFO does not mention (Reported by Alexander Traud) |
|
|
app_confbridge: Add support for disabling text messaging for a user (Reported by Joshua C. Colp) |
|
|
pjproject_bundled: Honor –without-pjproject. (Reported by Alexander Traud) |
|
|
res_rtp_asterisk: Loop when receive buffer is flushed by a received packet that is also in receive buffer with NACK (Reported by nappsoft) |
|
|
chan_sip: only sets ToS bits on UDP socket, ignoring TCP and TLS sockets (Reported by Joshua Roys) |
|
|
res_rtp_asterisk: Duplicate seqnos being added to send buffer with NACK (Reported by nappsoft) |
|
|
First DTMF is not get (Reported by Bernard Merindol) |
|
|
pjsip startup errors when using “with-ssl” configure option (Reported by Patrick Wakano) |
|
|
BuildSystem: Search for Python/C API when possibly needed only. (Reported by Alexander Traud) |
|
|
BuildSystem: In NetBSD, the Python Programming Language is python-2.7. (Reported by Alexander Traud) |
|
|
chan_pjsip: constant DTMF tone if RTP is not setup yet (Reported by Kevin Harwell) |
|
|
bridge_softmix_binaural: Show state in menuselect. (Reported by Alexander Traud) |
|
|
BuildSystem: Remove doc/tex and doc/pdf leftovers. (Reported by Alexander Traud) |
|
|
BuildSystem: Allow space in path. (Reported by Alexander Traud) |
|
|
res_rtp_asterisk: Avoid absolute value on unsigned subtraction. (Reported by Alexander Traud) |
|
|
func_channel: cannot read fields exten, context, userfield, channame from dialplan (Reported by Sébastien Duthil) |
|
|
chan_unistim: Avoid tautological warnings with clang. (Reported by Alexander Traud) |
|
|
test_stasis: Avoid always true warning with clang. (Reported by Alexander Traud) |
|
|
res_pjsip: Incorrect endpoint status after endpoint synchronization for a specific AOR (Reported by Jason Hord) |
|
|
channel: write to a stream on multi-frame writes (Reported by Kevin Harwell) |
|
|
test_utils: incorrectly printing error ‘declined to load’ (Reported by Alexander Traud) |
|
|
func_aes: incorrectly printing error ‘declined to load’ (Reported by Alexander Traud) |
|
|
Crash during conference call using confbridge and video (Reported by Pascal Cadotte Michaud) |
|
|
DAHDIRAS fails to properly initiate pppd unless asterisk is running as root (Reported by Jaco Kroon) |
|
|
dundi_read_result crash due to negative number (Reported by Jaco Kroon) |
|
|
res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream (Reported by Joshua C. Colp) |
|
|
Asterisk is crashing if the 200 OK with SDP (Reported by sungtae kim) |
|
|
res_pjsip_session: Allow default non-audio streams to have reflected state (Reported by Joshua C. Colp) |
|
|
chan_pjsip’s rtptimeout is erroneously triggered during direct-media (native_rtp) bridge (Reported by Michael Neuhauser) |
|
|
Comments in configs/func_odbc.conf.sample are not consistent with examples. Missing examples. (Reported by Olivier Krief) |
|
|
app_mixmonitor: Memory leak due to race condition between AMI MixMonitor and hangup (Reported by Joshua C. Colp) |
|
|
Incorrect Sender SSRC in RTCP when p2p rtp bridge is active (Reported by Torrey Searle) |
|
|
DTLS Handshake Fails to Occur if ice_support is enabled but not used (Reported by Torrey Searle) |
|
|
A non negotiated rtp frame causes call disconnection when there is a SSRC change (Reported by Paulo Vicentini) |
|
|
func_enum: ENUM code wrong case (Reported by Vitold) |
|
|
Fix the FSF address in the headers of lots of pjproject files (Reported by Jared Smith) |
|
|
Function TXTCIDNAME never actually makes DNS calls and always returns an empty string (Reported by George Joseph) |
|
|
PJSIP blind transfer not completed after using Proceeding() (Reported by lvl) |
|
|
res_rtp_asterisk: Improve NACK support and seqno handling (Reported by Joshua C. Colp) |
|
|
SIP/Stasis: SIP headers not transmitted in the “variables” field (Reported by Jean Aunis – Prescom) |
|
|
check_expr2: linking (when hardening) and cross-compiling troubles (Reported by Sebastian Kemper) |
|
|
ASTERISK-28738 Causes Audio Issue After Hold (Reported by Ross Beer) |
|
|
res_pjsip: Named ACL does not update on reload if changed (Reported by Timothy Vanderaerden) |
|
|
res_pjsip_outbound_ (Reported by George Joseph) |
|
|
ICE: pjnath shouldn’t wait for ICE to complete before allowing sending (Reported by Benjamin Keith Ford) |
|
|
Incorrect state machine used when MOH_PASSTHRU is used (Reported by Torrey Searle) |
|
|
res_rtp_asterisk: static for audio due to incomplete dtls/srtp setup (Reported by Kevin Harwell) |
|
|
Realtime MoH Unknown format ” — defaulting to SLIN (Reported by Ross Beer) |
|
|
res_pjsip_session: Fix out of order session refreshes (Reported by Joshua C. Colp) |
|
|
pjsip: SIP Packets with Via “received=” Containing IPv6 Address Delimited by “[]” Rejected (Reported by Peter Sokolov) |
|
|
chan_sip: Returns 403 if RTP ports are depleted, should return 503 (Reported by Walter Doekes) |
|
|
res_stasis_playback: Error building JSON (Reported by Sébastien Duthil) |
|
|
REGRESSION: Feature subscription_persistence_ (Reported by Ross Beer) |
|
|
res_pjsip_messaging: MessageSend Content-Type can’t be changed (Reported by Alex) |
|
|
ARI causes STASIS Deadlock (Reported by Ross Beer) |
|
|
stasis application is destroyed after its creation (Reported by Francois Blackburn) |
|
|
PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the error when sending (Reported by Dmitriy Serov) |
|
|
chan_sip strictrtp=yes fails when media source is changed: no audio (Reported by Walter Doekes) |
|
|
RTP Stream Incorrect Payload Type Causes Asterisk To Drop Calls (Reported by Paul Brooks) |
|
|
CDR billsec is always 0 for transferred calls (Reported by Maciej Michno) |
|
|
chan_dahdi: holding a channel via flash to dialtone times out after 0:16:40 (Reported by Andrew Siplas) |
|
|
Update documentation for statsd module – usage requirements unclear (Reported by Dan Jenkins) |
|
|
silk 24hHz doesn’t show up in ‘core show translation’ output (Reported by Sean Bright) |
|
|
core: minmemfree watermark uses free RAM, not available RAM (Reported by Kevin Flyn) |
|
|
chan_sip: SIP MESSAGE beginning with a whitespace appears empty in the dialplan (Reported by Frank Matano) |
|
|
Segfault forwarding voicemail with ODBC storage enabled and realtime voicemail_data is used (Reported by Stas Kobzar) |
|
|
empty voicemail.conf required for ARA (realtime) voicemail to leave message (Reported by Jim Van Meggelen) |
|
|
CLI command ‘realtime update2’ syntax failure when using according to usage help (Reported by Cedric BASSAGET) |
|
|
Pause reason not reported in QueueMember AMI event (Reported by Niksa Baldun) |
|
|
res_pjsip_endpoint_identifier_ (Reported by Joshua C. Colp) |
|
|
res_pjsip_notify: Multiple Event headers can be present instead of just one (Reported by AvayaXAsterisk) |
|
|
app_record: Lack of `beep` audio file causes application to return error and hangup (Reported by Corey Farrell) |
|
|
Wiki docs missing for MessageWaiting (Reported by David M. Lee) |
|
|
res_pjsip_pubsub: Subscription persistence does not preserve XML version number (Reported by Bryan Nelson) |
|
|
chan_dahdi: Deadlock in Hangup Scenarios with concurrent command pri show span X (Reported by Dirk Wendland) |
|
|
stasis bridge topic leak (Reported by Joeran Vinzens) |
|
|
pjsip reload not reloading wizard endpoint/pickup_group endpoint/call_group (Reported by Jean-Denis Girard) |
|
|
SIP WSS message not processed until next frame arrives (Reported by Robert Sutton) |
|
|
Asterisk ignores parsing of config files if a Byte order mark is present (Reported by Robin Leffmann) |
|
|
Playback of local files impacted by large media cache (Reported by Kevin Reeves) |
|
|
contrib: valgrind.supp doesn’t suppress what it’s supposed to due to invalid syntax (Reported by Richard Kenner) |
|
|
“trustrpid” is misspelled in sip_to_pjsip.py (Reported by Pascal Cadotte Michaud) |
|
|
app_chanisavail+cdr: ChanIsAvail sometimes fails to deactivate CDR. (Reported by Frederic LE FOLL) |
|
|
app_meetme, chan_ooh323 and cdr_mysql don’t build on 17.0.0 (Reported by George Joseph) |
|
|
res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them (Reported by nappsoft) |
|
|
res_fax: wrap Asterisk initiated negotiation with config option (Reported by Kevin Harwell) |
|
|
Missing arguments in PJSIP_CONTACT function documentation (Reported by Pascal Cadotte Michaud) |
|
|
Memory Leak in res_rtp_asterisk.c (Reported by Ted G) |
|
|
chan_sip logs errors on tx to non-existent TCP connections (Reported by Jaco Kroon) |
|
|
chan_pjsip incorrectly re-writes REGISTER 200 Response Contact (Reported by Ross Beer) |
|
|
res_pjsip Segfaults when realtime configuration to an AOR points to a not existent AOR (Reported by Ross Beer) |
|
|
chan_sip: RTP frames not transmitted after emitting a COLP (Reported by Jean Aunis – Prescom) |
|
|
chan_sip+native_bridge_rtp: directmedia compatibility check failure when negociated ptime is not default ptime. (Reported by Frederic LE FOLL) |
|
|
res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on hangup when TEST_FRAMEWORK enabled (Reported by Bernhard Schmidt) |
|
|
res_parking: Doesn’t park when parkee and parker are the same (Reported by Ross Beer) |
|
|
Enforce T.38 error correction mode at 200 ok received (Reported by Salah Ahmed) |
|
|
res_pjsip_outbound_ (Reported by Kevin Harwell) |
|
|
app_amd: Use time calculation to calculate timeout (Reported by Michael Cargile) |
|
|
chan_dahdi: PRI span status may stay “Down, Active” after a short alarm (Reported by Frederic LE FOLL) |
|
|
res_rtp_asterisk: ICE Completion Crash when sent packet length doesn’t match (Reported by Joshua Elson) |
|
|
FILE function grabs garbage along with read data when target line has no newline (Reported by Jonathan Harris) |
|
|
bridge_softmix: hold not cleared when joining a softmix bridge (Reported by Kevin Harwell) |
|
|
parking: Deadlock when multi call parking (Reported by Joshua C. Colp) |
|
|
Memory leaks in res_calendar_exchange and res_calendar_icalendar (Reported by Yoooooo Ha) |
|
|
ari/resource_events: Crash in event session cleanup (Reported by Kevin Harwell) |
|
|
utils.c throws repeated warnings; “pthread_attr_setstacksize: Invalid argument” (Reported by Speed Dial Dave) |
|
|
race condition on pjsip channelstats command (Reported by Salah Ahmed) |
|
|
cdr_pgsql: accesses obsolete (and finally removed) column (Reported by Christoph Moench-Tegeder) |
|
|
MWI Send Notify Crash on 16.6 (Reported by Joshua Elson) |
|
|
pjproject fails to build on 16.6.0, works on 16.5 (Reported by Niklas Larsson) |
|
|
Asterisk Deadlocks (Reported by Aheliotech) |
|
|
chan_pjsip: Crash when initiating PlayDTMF over AMI (Reported by Jeremiah Gadd) |
|
|
res_pjsip_mwi: Frack during unload on unsolicited_mwi container (Reported by Kevin Harwell) |
|
|
CDR backend unload problem during active call(s) (Reported by Marian Piater) |
|
|
stasis.c: Crash during unload (Reported by Kevin Harwell) |
|
|
Wrong contact representation in ipv6 mode (Reported by Jørgen H) |
|
|
Segmentation fault when there is no priority for an extension (Reported by Timothy Vanderaerden) |
|
|
res_pjsip_path: Crash when invalid contact is configured (Reported by Juan Martin) |
|
|
pjsip: Memory Leak (Reported by Mark) |
|
|
Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière) |
|
|
Asterisk release candidates fail to build on FreeBSD (Reported by Guido Falsi) |
|
|
chan_pjsip: Deadlock on fax detection (Reported by Joshua C. Colp) |
|
|
func_odbc: truncating Unicode string on readsql (Reported by Boris P. Korzun) |
|
|
setvar directive when used in template and a child of said template, results in duplicate variable names (Reported by Michael Goryainov) |
|
|
ChanIsAvail() creates a CDR if unanswered=yes is set in cdr.conf (Reported by Frederic LE FOLL) |
|
|
chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up (Reported by Frederic LE FOLL) |
|
|
codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32 (Reported by Ruddy G) |
|
|
translate: Crash when frame does not have a “src” field set (Reported by Gregory Massel) |
|
|
chan_unistim: Clang Warning: variable sized type not at end of a struct (Reported by Alexander Traud) |
|
|
pjsip mwi: n+1 sip notify’s sent on re-register (Reported by Chris Savinovich) |
|
|
PJSIP cnonce generated on Linux contains 36 characters, NEC only supports up to 32 characters (Reported by Dan Cropp) |
|
|
app_voicemail/IMAP: segfault in leave_voicemail because not checking mailstream (Reported by Alexei Gradinari) |
|
|
compile menuselect on gentoo (Reported by Kilburn) |
|
|
Asterisk occasionally passes a NULL as srtp->session to srtp_protect/unprotect causing SEGV (Reported by Jonas Swiatek) |
|
|
cel / cdr: Event times may be incorrect (Reported by Joshua C. Colp) |
|
|
json integer overflow in ssrc and timestamp (Reported by Salah Ahmed) |
|
|
res_pjsip: pjsip show contacts prints double entries (Reported by Ian Jones) |
|
|
packet lost on UDPTL wrap around (Reported by Torrey Searle) |
Improvements made in this release:
——————————
|
res_pjsip: Added option for disable rport parameter set (Reported by sungtae kim) |
|
|
Continue reading string when ping received by websocket (Reported by Nickolay V. Shmyrev) |
|
|
AMI SendText – add Content-Type parameter (Reported by Kevin Harwell) |
|
|
res_http_websocket: Add masking to websocket client (Reported by Moises Silva) |
|
|
Upgrade Asterisk to bundled pjproject 2.10 (Reported by Kevin Harwell) |
|
|
res_pjsip_logger: Add tons’o’functionality (Reported by Joshua C. Colp) |
|
|
ari: Add support for specifying variables on channel create (Reported by Joshua C. Colp) |
|
|
pjproject has race conditions in it’s build system (Reported by Guido Falsi) |
|
|
third-party/pjproject/ (Reported by Guido Falsi) |
|
|
Missing include on FreeBSD (Reported by Guido Falsi) |
|
|
chan_mobile creates PCMA streams that make some VoIP clients crash or not render received audio (Reported by Peter Turczak) |
|
|
func_volume: Allow decimal numbers as parameter to improve granularity (Reported by Jean Aunis – Prescom) |
|
|
Codec Negotiation: add outgoing_call_offer_prefs option (Reported by Kevin Harwell) |
|
|
dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn’t (Reported by Joshua Elson) |
|
|
Add support for Content-Disposition header in multi-part INVITES (Reported by Torrey Searle) |
|
|
res_pjsip_session: Decide more intelligently when to add video (Reported by Joshua C. Colp) |
|
|
Codec Negotiation: add incoming_call_offer_pref option (Reported by Kevin Harwell) |
|
|
TLS/SSL Key too small error (Reported by Martin Zeh) |
|
|
stream: Add support for adding/removing streams during SFU/calls (Reported by Joshua C. Colp) |
|
|
Documentation – Clarify That Format Is Set By File Name Extension In MixMonitor (Reported by xrobau) |
|
|
install_prereq script uses the interactive mode when installing aptitude (Reported by Sylvain Afchain) |
|
|
Should be able to disable the /httpstatus URI in the built-in HTTP server (Reported by Sean Bright) |
|
|
Add AudioSocket support (Reported by Seán C. McCord) |
|
|
Simplify dialplan for Dial, Page, and ChanIsAvail (Reported by cmaj) |
|
|
GET FULL VARIABLE documentation clarification (Reported by Jonathan Harris) |
|
|
Add an “inhibitCOLP” flag to the bridges REST API (Reported by Jean Aunis – Prescom) |
|
|
app_confbridge: Add support for setting maximum sample rate (Reported by Joshua C. Colp) |
|
|
res_pjsip_outbound_ (Reported by Daniel) |
|
|
Typo in README-SERIOUSLY. (Reported by Sam Banks) |
|
|
Allow voicemail forwards with ODBC backend when format differs from attachfmt column (Reported by cmaj) |
|
|
Problem with ASTERISK-20207: Asterisk should clear out any .lock files in the voice mail directory on startup. (Reported by Michael) |
|
|
add the ability for asterisk to generate on-hold re-invites (Reported by Torrey Searle) |
|
|
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.
Thank you for your continued support of Asterisk!