The Asterisk Development Team would like to announce the first release candidate of Asterisk 19.0.0. This release candidate is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 19.0.0-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: Deprecations made in this release: ----------------------------------- * ASTERISK-29601 - moduleinfo: Add replacement module information (Reported by N A) * ASTERISK-29602 - res_monitor: Disable building by default. (Reported by Joshua C. Colp) * ASTERISK-29600 - muted: Remove deprecated application (Reported by Joshua C. Colp) * ASTERISK-29599 - conf2ael: Remove deprecated application (Reported by Joshua C. Colp) * ASTERISK-29598 - res_config_sqlite: Remove deprecated module (Reported by Joshua C. Colp) * ASTERISK-29597 - chan_vpb: Remove deprecated module (Reported by Joshua C. Colp) * ASTERISK-29596 - chan_misdn: Remove deprecated module (Reported by Joshua C. Colp) * ASTERISK-29595 - chan_nbs: Remove deprecated module (Reported by Joshua C. Colp) * ASTERISK-29594 - chan_phone: Remove deprecated module (Reported by Joshua C. Colp) * ASTERISK-29593 - chan_oss: Remove deprecated module (Reported by Joshua C. Colp) * ASTERISK-29592 - cdr_syslog: Remove deprecated module (Reported by Joshua C. Colp) * ASTERISK-29591 - app_dahdiras: Remove deprecated module (Reported by Joshua C. Colp) * ASTERISK-29590 - app_nbscat: Remove deprecated module (Reported by Joshua C. Colp) * ASTERISK-29589 - app_image: Remove deprecated module (Reported by Joshua C. Colp) * ASTERISK-29588 - app_url: Remove deprecated module (Reported by Joshua C. Colp) * ASTERISK-29587 - app_fax: Remove deprecated module (Reported by Joshua C. Colp) * ASTERISK-29586 - app_ices: Remove deprecated module (Reported by Joshua C. Colp) * ASTERISK-29585 - app_mysql: Remove deprecated module (Reported by Joshua C. Colp) * ASTERISK-29584 - cdr_mysql: Remove deprecated module (Reported by Joshua C. Colp) * ASTERISK-29548 - app_meetme: Deprecated in 19, to be removed in 21 (Reported by Joshua C. Colp) * ASTERISK-29549 - app_osploop: Deprecated in 19, to be removed in 21 (Reported by Joshua C. Colp) * ASTERISK-29550 - chan_alsa: Deprecated in 19, to be removed in 21 (Reported by Joshua C. Colp) * ASTERISK-29551 - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported by Joshua C. Colp) * ASTERISK-29552 - chan_skinny: Deprecated in 19, to be removed in 21 (Reported by Joshua C. Colp) * ASTERISK-29553 - res_pktccops: Deprecated in 19, to be removed in 21 (Reported by Joshua C. Colp) * ASTERISK-29558 - app_macro: Deprecated in 16, to be removed in 21 (Reported by Joshua C. Colp) * ASTERISK-29567 - chan_sip: Deprecated in 17, to be removed in 21 (Reported by Joshua C. Colp) * ASTERISK-29572 - res_monitor: Deprecated in 16, to be removed in 21 (Reported by Joshua C. Colp) Security bugs fixed in this release: ----------------------------------- * ASTERISK-29381 - chan_pjsip: Remote denial of service by an authenticated user (Reported by Ivan Poddubny) * ASTERISK-29415 - Crash in PJSIP TLS transport (Reported by Andrew Yager) * ASTERISK-29305 - ASTERISK-29203 / AST-2021-002 -- Another scenario is causing a crash (Reported by Gregory Massel) * ASTERISK-29260 - sRTP Replay Protection ignored; even tears down long calls (Reported by Alexander Traud) * ASTERISK-29227 - res_pjsip_diversion: sending multiple 181 responses causes memory corruption and crash (Reported by Ivan Poddubny) * ASTERISK-29219 - res_pjsip_diversion: Crash if Tel URI contains History-Info (Reported by Torrey Searle) * ASTERISK-29057 - pjsip: Crash on call rejection during high load (Reported by Sandro Gauci) New Features made in this release: ----------------------------------- * ASTERISK-29656 - Add CHANNEL_EXISTS function (Reported by N A) * ASTERISK-29496 - Add SendMF application (Reported by N A) * ASTERISK-29627 - Add STRBETWEEN function (Reported by N A) * ASTERISK-29628 - Add file and directory functions (Reported by N A) * ASTERISK-29531 - Add SAYFILES function (Reported by N A) * ASTERISK-29546 - Add tone detection module (Reported by N A) * ASTERISK-18454 - Option for Read to be able to accept # (Reported by Sta Retji) * ASTERISK-29542 - Add audio scrambler (Reported by N A) * ASTERISK-29478 - Function to drop frames in the TX or RX directions (Reported by N A) * ASTERISK-29389 - Add PJSIP_HEADERS() and ability to read header by pattern (Reported by Igor Goncharovsky) * ASTERISK-11 - AGI channel_status failure (Reported by bbawkon) * ASTERISK-29477 - Function to asynchronously store digits dialed (Reported by N A) * ASTERISK-29454 - New application to reload modules (Reported by N A) * ASTERISK-29444 - Add application to wait for condition (Reported by N A) * ASTERISK-29442 - app_dial: Expand A option to allow announcement playback to caller (Reported by N A) * ASTERISK-29446 - app_confbridge: New ConfKick application (Reported by N A) * ASTERISK-29440 - app_confbridge: Allow ConfBridge answer to be suppressed (Reported by N A) * ASTERISK-29431 - Minimum and maximum dialplan functions (Reported by N A) * ASTERISK-29439 - func_volume: Volume function can't be read (Reported by N A) * ASTERISK-27477 - Chan_pjsip does not support unauthenticated OPTIONS ping (Reported by Ross Beer) * ASTERISK-29027 - Implement support for History-Info (Reported by Torrey Searle) Bugs fixed in this release: ----------------------------------- * ASTERISK-20219 - [patch] - IAX2 Call Encryption Fails with RSA authentication (Reported by Michael Munger) * ASTERISK-29402 - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform does not support it (Reported by Matthew Kern) * ASTERISK-29673 - app_read: Fix null pointer crash regression (Reported by N A) * ASTERISK-29671 - res_rtp_asterisk: memory leak (Reported by Jean Aunis - Prescom) * ASTERISK-29668 - ari: Listing bridges fails when dialing bridge exists (Reported by Joshua C. Colp) * ASTERISK-29663 - messaging: AMI MessageSend does not support same parameters as dialplan application (Reported by Brian J. Murrell) * ASTERISK-29578 - app_queue: Custom device state using included hints do not update (Reported by N A) * ASTERISK-29660 - Build failure when disabling PJSIP support (Reported by Guido Falsi) * ASTERISK-29654 - pjproject includes trailing whitespace in sdp format attributes (Reported by George Joseph) * ASTERISK-29635 - MP3Player don' t work with actual mpg123 versions (Reported by Carlos Oliva) * ASTERISK-29629 - ARI external media channel creation doesn't set option data (Reported by sungtae kim) * ASTERISK-27176 - test_abstract_jb: frames leak (Reported by Corey Farrell) * ASTERISK-29634 - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported by George Joseph) * ASTERISK-29630 - Asterisk is unable to read extended number format terminfo files (Reported by Sean Bright) * ASTERISK-28004 - dns: Core ast_dns_get_nameservers does not support configured IPv6 servers (Reported by Isaac McDonald) * ASTERISK-29618 - ConfBridge errors on creation conference room (Reported by Alexander Zharov) * ASTERISK-29622 - ARI: external media create doesn't use body parameter (Reported by sungtae kim) * ASTERISK-29614 - app_agent_pool: XML Doc: unterminated entity reference (Reported by Alexander Traud) * ASTERISK-29609 - Subsequent 'ael reload' will cause a lock up (Reported by Mark Murawski) * ASTERISK-28701 - app_queue: Core reload resets queue stats, even when keepstats=yes (Reported by Luke Escude) * ASTERISK-29616 - res_rtp_asterisk: sqrt(.) requires the header math.h. (Reported by Alexander Traud) * ASTERISK-29518 - sig_analog: FCG_CAMA fails to signal ANI spill when using MF signaling (Reported by Sarah Autumn) * ASTERISK-29582 - res_pjproject: Can't map pjproject log messages to Asterisk TRACE (Reported by George Joseph) * ASTERISK-29575 - app_milliwatt: Milliwatt application doesn't use the proper timings (Reported by N A) * ASTERISK-20339 - chan_mgcp, resp_pktccops ast_debug support (Reported by Tomas Maldonado) * ASTERISK-29540 - aelparse: include of context with timings fails (Reported by Alexander Traud) * ASTERISK-29539 - Segmentation fault at ast_writestream() when write handler not defined (happens with OGG/Speex) (Reported by Ernani José Camargo Azevedo) * ASTERISK-29494 - cdr_adaptive_odbc: Prevent throwing warnings if CDR filtering is used (Reported by N A) * ASTERISK-29513 - statsd: Remove non-standard metric type Meter (Reported by Rijnhard Hessel) * ASTERISK-12 - app_voicemail2 became a bit silent, lately (Reported by siggi) * ASTERISK-29526 - G729 audio gets corrupted by Asterisk due to smoother (Reported by under) * ASTERISK-29392 - chan_iax2: Asterisk crashes when queueing video with format (Reported by Michael Welk) * ASTERISK-27871 - Remote URL in playback must end with file extension (Reported by Caesar) * ASTERISK-29507 - STUN timeout is silently delaying calls (Reported by Sébastien Duthil) * ASTERISK-29514 - ari: Audiosocket segfault when no data specified (Reported by Igor Goncharovsky) * ASTERISK-29503 - Updated identify/match syntax not supported by config wizard (Reported by Sean Bright) * ASTERISK-29480 - fixedjitterbuffer contains an un-wrappered assert that triggers on a negative time slew (Reported by Dan Cropp) * ASTERISK-29485 - core: Inband generation of tones for Busy() and Congestion() may not occur (Reported by Joshua C. Colp) * ASTERISK-29479 - [patch] Channels are not put on hold for Session Progress with inactive audio (Reported by Bernd Zobl) * ASTERISK-29475 - SayNumber triggers WARNING if caller hangs up during application execution (Reported by N A) * ASTERISK-29404 - Consolidate res_pjsip_messaging fixes for domain name (Reported by George Joseph) * ASTERISK-29441 - Core reload making TCP endpoints go offline (Reported by Luke Escude) * ASTERISK-28237 - "FRACK!, Failed assertion bad magic number" happens when unsubscribe an application from an event source (Reported by Lucas Tardioli Silveira) * ASTERISK-28393 - Multidomain support issue (Reported by Andrea Sannucci) * ASTERISK-29433 - res_rtp_asterisk: Server reflexive candidates use incorrect raddr for RTCP (Reported by Chris) * ASTERISK-29397 - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by George Joseph) * ASTERISK-24601 - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY event: dialog XML body (Reported by Marco Paland) * ASTERISK-29372 - file.c switch does not account for flash events (Reported by N A) * ASTERISK-29370 - chan_sip does not recognize application/hook-flash (Reported by N A) * ASTERISK-29377 - cpool_release_pool "double free or corruption (out)" (Reported by Robert Sutton) * ASTERISK-29358 - chan_pjsip: Trace message for progress is output even if frame is not queued (Reported by Michael Maier) * ASTERISK-29030 - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC) gets inserted when switching from progress to established (Reported by Matthias Hensler) * ASTERISK-29407 - chan_local: Filtering audio formats should not occur on removed streams (Reported by Joshua C. Colp) * ASTERISK-29328 - translate.c: possible buffer overflow when upsampling (Reported by Jean Aunis - Prescom) * ASTERISK-29379 - Segfault - ast_channel_is_multistream (chan=0x0) at channel_internal_api.c:1590 (Reported by Ross Beer) * ASTERISK-29130 - prometheus: Crash when scraping bridge (Reported by Francisco Correia) * ASTERISK-29364 - res_rtp_asterisk: standard deviation miscalculation (Reported by Kevin Harwell) * ASTERISK-29373 - res_rtp_asterisk: Flash events are duplicated (Reported by N A) * ASTERISK-28356 - app_queue: CLI set ringinuse for realtime member not working (Reported by Michael) * ASTERISK-24434 - Fix differing usage of assignment operators in modules.conf (Reported by Rusty Newton) * ASTERISK-26614 - app_queue: updatecdr option in queues.conf does effectively nothing (Reported by Alexander Gonchiy) * ASTERISK-24631 - Incorrect description of option "context" in queues.conf.sample (Reported by Etienne Lessard) * ASTERISK-25358 - dateformat not read from logger.conf by remote console (Reported by Igor Liferenko) * ASTERISK-27542 - app_queue: When "queue show" CLI command is executed a crash occurs (Reported by Miguel Sanz) * ASTERISK-29215 - res_pjsip_session: NULL active_media_state topology caused asterisk crash (Reported by sungtae kim) * ASTERISK-29355 - app_queue: Queue member status message sent even if status doesn't change (Reported by Roman Pertsev) * ASTERISK-29035 - chan_local: Multistream support breaks T.38 faxing (Reported by Matthias Hensler) * ASTERISK-29354 - res_pjsip: Allow partial reloading of transports (Reported by Joshua C. Colp) * ASTERISK-29348 - menuselect doesn't return errors in many cases (Reported by George Joseph) * ASTERISK-29352 - res_rtp_asterisk: Fix frame delivery time when SSRC changes (Reported by Joshua C. Colp) * ASTERISK-29071 - app_confbridge: Memory rises when jitterbuffer enabled and muting over AMI occurs (Reported by Stefan Ruf) * ASTERISK-29329 - app_dial: DTMF to 'D' option gets duplicated if there are multiple progress events (Reported by N A) * ASTERISK-29306 - strings: Incorrect use of __attribute__((pure)) in ast_str_to_lower definition (Reported by Vitezslav Novy) * ASTERISK-29300 - res_rtp_asterisk: When native local bridging the remote SSRC becomes permanent (Reported by Sebastian Damm) * ASTERISK-29235 - res_pjsip_nat: Contact is rewritten on REGISTER responses with external_signaling_address (Reported by Brian Paboojian) * ASTERISK-29266 - ICE Role conflict with an unauthorized session (Reported by Salah Ahmed) * ASTERISK-29105 - chan_pjsip: 180 Ringing with SDP not changed into progress (Reported by Sebastian Damm) * ASTERISK-29297 - say: Y2021 problem â Asterisk cannot say year 2021 in Dutch (Reported by Jacek Konieczny) * ASTERISK-29315 - res_pjsip: re-registration gets stuck if setting initial auth credentials fails (Reported by Nick French) * ASTERISK-29312 - res_fax: asterisk fails to publish the Stasis and ReceiveFax status messages if the remote Station ID contains invalid UTF-8 characters (Reported by Alexei Gradinari) * ASTERISK-16799 - Callee declined when 'beep' audio file does not exist (Reported by IAMJames_) * ASTERISK-29313 - res_pjsip_refer: Segfault in progress notify (Reported by George Joseph) * ASTERISK-28452 - pjsip: <sess-version> of SDP is not incremented though SDP may be changed on reinvite without SDP offer (Reported by Michael Maier) * ASTERISK-29311 - res_odbc_transaction sets forcecommit default value based on isolation level instead of forcecommit (Reported by Jaco Kroon) * ASTERISK-29303 - pjsip: Re-invite occurs when it shouldn't (Reported by Benjamin Keith Ford) * ASTERISK-29293 - res_config_pgsql: Limit realtime_pgsql() to return one (no more) record (Reported by Boris P. Korzun) * ASTERISK-28369 - app_queue: Member device state "invalid" when second call is ringing and hint is used (Reported by Boolah ) * ASTERISK-29287 - app.h: C++ compatibility broken (Reported by Jean Aunis - Prescom) * ASTERISK-29203 - res_pjsip_t38: Crash when changing state (Reported by Gregory Massel) * ASTERISK-29205 - res_rtp_asterisk: Asterisk crashes when making hold/unhold from webrtc client (Reported by Edvin Vidmar) * ASTERISK-29196 - res_pjsip: Segmentation fault (Reported by Mauri de Souza Meneguzzo (3CPlus)) * ASTERISK-29280 - chan_sip: Allow peers without audio (text+video). (Reported by Alexander Traud) * ASTERISK-29265 - chan_sip: Allow text+video media streams, again. (Reported by Alexander Traud) * ASTERISK-29259 - channel: Allow text+video media streams, again. (Reported by Alexander Traud) * ASTERISK-29261 - res_pjsip: user=phone validation fail for isup numbers containing *# (Reported by Mark Petersen) * ASTERISK-29258 - chan_sip: Audio stream rejected, Other stream present: Invalid SDP. (Reported by Alexander Traud) * ASTERISK-29248 - res_pjsip_session: res sometimes uninitialized reported by compiler Clang. (Reported by Alexander Traud) * ASTERISK-29220 - After T38 reinvite response of 488 a subsequent G711 reinvite is not processed correctly. Instead the previous T38 session media is used (Reported by Robert Cripps) * ASTERISK-29229 - Stasis/messaging: text messages not dispatched to all subscribers when using generic subscription (Reported by Jean Aunis - Prescom) * ASTERISK-29238 - chan_sip: SDP: Offers without any enabled stream are accepted. (Reported by Alexander Traud) * ASTERISK-29237 - chan_sip: SDP: m=video is parsed even when disabled. (Reported by Alexander Traud) * ASTERISK-29222 - chan_sip: Hold/Resume an sRTP call on a video enabled user-agent. (Reported by Alexander Traud) * ASTERISK-29240 - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN instead of a channel variable (Reported by Ivan Poddubny) * ASTERISK-27902 - chan_pjsip isn't updating hangupcause on 4XX responses (Reported by George Joseph) * ASTERISK-28016 - PJSIP sends duplicate 183 Progress responses (Reported by Alex Hermann) * ASTERISK-28185 - chan_pjsip: Subsequent same responses are not stopped (Reported by Julien) * ASTERISK-29230 - pjsip: Asterisk goes crazy and massively spams logfile if registration can't be send (Reported by Michael Maier) * ASTERISK-29231 - pjsip: SIGSEGV in CLI if no trunk is registered (Reported by Michael Maier) * ASTERISK-29217 - LOCK() can grant the same lock to multiple channels spuriously (Reported by Jaco Kroon) * ASTERISK-28947 - Segmentation fault in mixmonitor_ds_destroy (Reported by Robert Sutton) * ASTERISK-29201 - Crash occurs when Transfer and execute Hangup before the Transfer result (Reported by Dan Cropp) * ASTERISK-29168 - Asterisk crashes during call transfer (Reported by Dalius Mockevicius) * ASTERISK-29210 - res_pjsip: Crash when examining transport (Reported by N GM ) * ASTERISK-29191 - tel: URI in Diversion header causes crash (Reported by Mikhail Ivanov) * ASTERISK-28883 - Spyee information ist missing in ChanSpyStop AMI Event (Reported by Hendrik Wedhorn) * ASTERISK-29188 - null media causing the Asterisk crash (Reported by sungtae kim) * ASTERISK-29209 - Debug messages printed by scope trace might be missing newlines (Reported by Alexander Traud) * ASTERISK-29024 - pjsip: Route Header in Cancel request incorrectly set (Reported by Flole Systems) * ASTERISK-29211 - res_musiconhold: Segfault on realtime music on hold without entries (Reported by Nathan Bruning) * ASTERISK-29022 - Crash when manipulating PJSIP invite dlg ref counts (Reported by Sean Bright) * ASTERISK-29173 - Media cache URL requests allow infinite redirects (Reported by Sean Bright) * ASTERISK-29175 - res_pjsip_stir_shaken: Fix module description (Reported by Stanislav Abramenkov) * ASTERISK-29148 - AST_MODULE_INFO no, MODULEINFO depend (Reported by Alexander Traud) * ASTERISK-29165 - res_pjsip: malformed header Accept-Encoding in OPTIONS response (Reported by Alexander Greiner-Baer) * ASTERISK-28798 - [patch] chan_sip: TCP/TLS client without server. (Reported by Alexander Traud) * ASTERISK-29161 - Incorrect setup of recall channels (Reported by Boris P. Korzun) * ASTERISK-29155 - app_queue: Deadlock between queues container and individual queues (Reported by George Joseph) * ASTERISK-28933 - res_pjsip.so fails to load when bundled pjproject is compiled without libssl (Reported by Walter Doekes) * ASTERISK-28825 - Any curl response checks out as valid even if 404 is returned. (Reported by dovid) * ASTERISK-29013 - res_pjsip: Asterisk doesn't stop sending invites (with auth) on 407 replies (Reported by Sebastian Damm) * ASTERISK-29142 - sip_to_pjsip.py: doesn't read globbed includes (Reported by Michael Newton) * ASTERISK-29144 - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander Traud) * ASTERISK-29146 - GCC Warnings: â%sâ directive argument is null. (Reported by Alexander Traud) * ASTERISK-29145 - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander Traud) * ASTERISK-29124 - res_pjsip: flow transport broken for outbound requests (Reported by Nick French) * ASTERISK-29136 - config: Sample features.conf incorrectly includes " around sound files (Reported by Benjamin M.) * ASTERISK-29123 - logger.conf.sample missing comment mark on line 115 (Reported by Andrew Siplas) * ASTERISK-29109 - res_pjsip_session: Asterisk 18 does not progress calls due to codec negotiation after upgrading from Asterisk 16 (Reported by Ross Beer) * ASTERISK-28430 - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF (Reported by under) * ASTERISK-29108 - resource_endpoints.c : Memory leak if endpoint not found (Reported by Jean Aunis - Prescom) * ASTERISK-29097 - res_pjsip_config_wizard: Crash when freeing string when failing to add extension (Reported by Vieri) * ASTERISK-26424 - app_voicemail: Undocumented behavior from VMSayName (Reported by Eric Smith) * ASTERISK-29051 - res_pjsip_sdp_rtp: Does not set correct values on RTP instance when "auto" DTMF is used (Reported by Sebastian Damm) * ASTERISK-29099 - res_musiconhold: Realtime MOH only loads a single entry (Reported by laszlovl) * ASTERISK-29091 - Crash when ast_translator_build_path fails (Reported by Jasper van der Neut) * ASTERISK-28311 - dsp: ast_dsp_silence_noise_with_energy wrong judgment of frame format (Reported by å¨å®¶å»º) * ASTERISK-29085 - func_curl: Segmentation fault when using CURL after setting httpheader CURLOPT (Reported by Péter Juhász) * ASTERISK-24329 - Music On Hold announcement cuts intro of music the first time it is played (Reported by Thomas Frederiksen) * ASTERISK-29089 - RTP Ports not cleared after hangup (Reported by Ross Beer) * ASTERISK-29081 - res_stasis: Add compare function for bridges moh container (Reported by Hajek Michal) * ASTERISK-28416 - Unable to get rtp codec payload code for slin (Reported by Brian J. Murrell) * ASTERISK-29014 - res_pjsip_session: Re-INVITE collisions aren't handled correctly (Reported by George Joseph) * 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-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-29034 - Lastpause of realtime members is reseting (Reported by Evandro César Arruda) * 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) Improvements made in this release: ----------------------------------- * ASTERISK-29637 - Add support for future dates in Say.c (Reported by Shloime Rosenblum) * ASTERISK-29525 - PJSIP remove_existing unavailable contacts (Reported by Joseph Nadiv) * ASTERISK-29661 - func_vmcount: Add support for multiple mailboxes (Reported by N A) * ASTERISK-29275 - Support of MIME-type for wav16 (Reported by Boris P. Korzun) * ASTERISK-29529 - Add custom logging level (Reported by N A) * ASTERISK-29472 - res_pjsip: OLI/ANI2 support missing (Reported by N A) * ASTERISK-29626 - app_stack: Include calling location if attempting to branch to nonexistent location (Reported by N A) * ASTERISK-29632 - Add option to Application_VoiceMail to suppress instructions only when a custom greeting is present (Reported by Charlie Smurthwaite) * ASTERISK-29605 - chan_iax2: Add ANI2 (Reported by N A) * ASTERISK-29508 - STUN server address refresh (Reported by Sébastien Duthil) * ASTERISK-29612 - bridge_basic: Don't throw warning if attended transfer is cancelled (Reported by N A) * ASTERISK-29544 - Media Cache - Delayed remote sound file retrieve delays all playbacks (Reported by Andre Barbosa) * ASTERISK-29495 - Return integer instead of float if response is a whole number (Reported by N A) * ASTERISK-29541 - app_morsecode: Add American Morse code (Reported by N A) * ASTERISK-29543 - app_originate: Allow specifying codec(s) to use (Reported by N A) * ASTERISK-29528 - Add support for multiple files for agent announcements (Reported by N A) * ASTERISK-29527 - res_http_media_cache: Cleanup audio format lookup in HTTP requests (Reported by Sean Bright) * ASTERISK-29501 - ARI - Stasis Playback doesn't hangup call when processing a list of invalid files (Reported by Andre Barbosa) * ASTERISK-29464 - ARI - PlaybackFinish skip error events (Reported by Andre Barbosa) * ASTERISK-29450 - Allow setting channel variables using Originate application (Reported by N A) * ASTERISK-29460 - Recognize application/hook-flash in PJSIP (Reported by N A) * ASTERISK-29459 - Missing configuration from PJSIP to SIP conversion script (Reported by N A) * ASTERISK-29434 - Asterisk reveals pjproject version in STUN packets (Reported by Jeremy Lainé) * ASTERISK-29380 - Add Flash AMI event to handle flash events (Reported by N A) * ASTERISK-29349 - Silent voicemail option is not completely silent (Reported by N A) * ASTERISK-29339 - loader: Let's output warnings for deprecated modules! (Reported by Joshua C. Colp) * ASTERISK-29337 - menuselect: Add ability to set deprecated in and removed in versions for modules (Reported by Joshua C. Colp) * ASTERISK-29335 - xml: Embed module information into core XML documentation. (Reported by Joshua C. Colp) * ASTERISK-29336 - documentation: Fix inconsistent support levels (Reported by Joshua C. Colp) * ASTERISK-29321 - sorcery: Add support for more intelligent reloading. (Reported by Joshua C. Colp) * ASTERISK-29325 - res_pjsip_registrar: Include source IP address and port in log messages (Reported by Joshua C. Colp) * ASTERISK-29326 - asterisk: Update copyright/company (Reported by Joshua C. Colp) * ASTERISK-29244 - Add MixMonitorStart / Stop / Mute AMI events (Reported by Sébastien Duthil) * ASTERISK-29252 - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER) failure SIP code (Reported by Dan Cropp) * ASTERISK-29262 - Support of various URL-schemes by MoH (Reported by Boris P. Korzun) * ASTERISK-28549 - Two repeated 183 (Reported by Gant Liu) * ASTERISK-29216 - contrib: systemd asterisk service for centos8 or other newer linux versions (Reported by Mark Petersen) * ASTERISK-29143 - res_http_media_cache: HTTP media cache stored hardcoded in /tmp (Reported by laszlovl) * ASTERISK-29118 - VoiceMail() should have an option to play greetings as Early Media (Reported by Juan Carlos Castro y Castro) * ASTERISK-29054 - Logger: Add debug logging categories (Reported by Kevin Harwell) * ASTERISK-29083 - Do not build chan_sip by default as it is now deprecated (Reported by Sean Bright) * ASTERISK-29056 - Increase reg_server column size for ps_contacts table realtime (Reported by sungtae kim) * ASTERISK-29055 - Create a Bridge with video_single mode (Reported by sungtae kim) For a full list of changes in this release candidate, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0-rc1 Thank you for your continued support of Asterisk!
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