The release of Asterisk 19.0.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Deprecations made in this release:
-----------------------------------
| moduleinfo: Add replacement module information | (Reported by N A) | |
| res_monitor: Disable building by default. | (Reported by Joshua C. Colp) | |
| muted: Remove deprecated application | (Reported by Joshua C. Colp) | |
| conf2ael: Remove deprecated application | (Reported by Joshua C. Colp) | |
| res_config_sqlite: Remove deprecated module | (Reported by Joshua C. Colp) | |
| chan_vpb: Remove deprecated module | (Reported by Joshua C. Colp) | |
| chan_misdn: Remove deprecated module | (Reported by Joshua C. Colp) | |
| chan_nbs: Remove deprecated module | (Reported by Joshua C. Colp) | |
| chan_phone: Remove deprecated module | (Reported by Joshua C. Colp) | |
| chan_oss: Remove deprecated module | (Reported by Joshua C. Colp) | |
| cdr_syslog: Remove deprecated module | (Reported by Joshua C. Colp) | |
| app_dahdiras: Remove deprecated module | (Reported by Joshua C. Colp) | |
| app_nbscat: Remove deprecated module | (Reported by Joshua C. Colp) | |
| app_image: Remove deprecated module | (Reported by Joshua C. Colp) | |
| app_url: Remove deprecated module | (Reported by Joshua C. Colp) | |
| app_fax: Remove deprecated module | (Reported by Joshua C. Colp) | |
| app_ices: Remove deprecated module | (Reported by Joshua C. Colp) | |
| app_mysql: Remove deprecated module | (Reported by Joshua C. Colp) | |
| cdr_mysql: Remove deprecated module | (Reported by Joshua C. Colp) | |
| app_meetme: Deprecated in 19, to be removed in 21 | (Reported by Joshua C. Colp) | |
| app_osploop: Deprecated in 19, to be removed in 21 | (Reported by Joshua C. Colp) | |
| chan_alsa: Deprecated in 19, to be removed in 21 | (Reported by Joshua C. Colp) | |
| chan_mgcp: Deprecated in 19, to be removed in 21 | (Reported by Joshua C. Colp) | |
| chan_skinny: Deprecated in 19, to be removed in 21 | (Reported by Joshua C. Colp) | |
| res_pktccops: Deprecated in 19, to be removed in 21 | (Reported by Joshua C. Colp) | |
| app_macro: Deprecated in 16, to be removed in 21 | (Reported by Joshua C. Colp) | |
| chan_sip: Deprecated in 17, to be removed in 21 | (Reported by Joshua C. Colp) | |
| res_monitor: Deprecated in 16, to be removed in 21 | (Reported by Joshua C. Colp) |
Security bugs fixed in this release:
-----------------------------------
| chan_pjsip: Remote denial of service by an authenticated user | (Reported by Ivan Poddubny) | |
| Crash in PJSIP TLS transport | (Reported by Andrew Yager) | |
| ASTERISK-29203 / AST-2021-002 -- Another scenario is causing a crash | (Reported by Gregory Massel) | |
| sRTP Replay Protection ignored; even tears down long calls | (Reported by Alexander Traud) | |
| res_pjsip_diversion: sending multiple 181 responses causes memory corruption and crash | (Reported by Ivan Poddubny) | |
| res_pjsip_diversion: Crash if Tel URI contains History-Info | (Reported by Torrey Searle) | |
| pjsip: Crash on call rejection during high load | (Reported by Sandro Gauci) |
New Features made in this release:
-----------------------------------
| Add CHANNEL_EXISTS function | (Reported by N A) | |
| Add SendMF application | (Reported by N A) | |
| Add STRBETWEEN function | (Reported by N A) | |
| Add file and directory functions | (Reported by N A) | |
| Add SAYFILES function | (Reported by N A) | |
| Add tone detection module | (Reported by N A) | |
| Option for Read to be able to accept # | (Reported by Sta Retji) | |
| Add audio scrambler | (Reported by N A) | |
| Function to drop frames in the TX or RX directions | (Reported by N A) | |
| Add PJSIP_HEADERS() and ability to read header by pattern | (Reported by Igor Goncharovsky) | |
| AGI channel_status failure | (Reported by bbawkon) | |
| Function to asynchronously store digits dialed | (Reported by N A) | |
| New application to reload modules | (Reported by N A) | |
| Add application to wait for condition | (Reported by N A) | |
| app_dial: Expand A option to allow announcement playback to caller | (Reported by N A) | |
| app_confbridge: New ConfKick application | (Reported by N A) | |
| app_confbridge: Allow ConfBridge answer to be suppressed | (Reported by N A) | |
| Minimum and maximum dialplan functions | (Reported by N A) | |
| func_volume: Volume function can't be read | (Reported by N A) | |
| Chan_pjsip does not support unauthenticated OPTIONS ping | (Reported by Ross Beer) | |
| Implement support for History-Info | (Reported by Torrey Searle) |
Bugs fixed in this release:
-----------------------------------
| [patch] - IAX2 Call Encryption Fails with RSA authentication | (Reported by Michael Munger) | |
| res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform does not support it | (Reported by Matthew Kern) | |
| app_read: Fix null pointer crash regression | (Reported by N A) | |
| res_rtp_asterisk: memory leak | (Reported by Jean Aunis - Prescom) | |
| ari: Listing bridges fails when dialing bridge exists | (Reported by Joshua C. Colp) | |
| messaging: AMI MessageSend does not support same parameters as dialplan application | (Reported by Brian J. Murrell) | |
| app_queue: Custom device state using included hints do not update | (Reported by N A) | |
| Build failure when disabling PJSIP support | (Reported by Guido Falsi) | |
| pjproject includes trailing whitespace in sdp format attributes | (Reported by George Joseph) | |
| MP3Player don' t work with actual mpg123 versions | (Reported by Carlos Oliva) | |
| ARI external media channel creation doesn't set option data | (Reported by sungtae kim) | |
| test_abstract_jb: frames leak | (Reported by Corey Farrell) | |
| res_snmp: gcc 11 needs -fPIC to compile correctly | (Reported by George Joseph) | |
| Asterisk is unable to read extended number format terminfo files | (Reported by Sean Bright) | |
| dns: Core ast_dns_get_nameservers does not support configured IPv6 servers | (Reported by Isaac McDonald) | |
| ConfBridge errors on creation conference room | (Reported by Alexander Zharov) | |
| ARI: external media create doesn't use body parameter | (Reported by sungtae kim) | |
| app_agent_pool: XML Doc: unterminated entity reference | (Reported by Alexander Traud) | |
| Subsequent 'ael reload' will cause a lock up | (Reported by Mark Murawski) | |
| app_queue: Core reload resets queue stats, even when keepstats=yes | (Reported by Luke Escude) | |
| res_rtp_asterisk: sqrt(.) requires the header math.h. | (Reported by Alexander Traud) | |
| sig_analog: FCG_CAMA fails to signal ANI spill when using MF signaling | (Reported by Sarah Autumn) | |
| res_pjproject: Can't map pjproject log messages to Asterisk TRACE | (Reported by George Joseph) | |
| app_milliwatt: Milliwatt application doesn't use the proper timings | (Reported by N A) | |
| chan_mgcp, resp_pktccops ast_debug support | (Reported by Tomas Maldonado) | |
| aelparse: include of context with timings fails | (Reported by Alexander Traud) | |
| Segmentation fault at ast_writestream() when write handler not defined (happens with OGG/Speex) | (Reported by Ernani José Camargo Azevedo) | |
| cdr_adaptive_odbc: Prevent throwing warnings if CDR filtering is used | (Reported by N A) | |
| statsd: Remove non-standard metric type Meter | (Reported by Rijnhard Hessel) | |
| app_voicemail2 became a bit silent, lately | (Reported by siggi) | |
| G729 audio gets corrupted by Asterisk due to smoother | (Reported by under) | |
| chan_iax2: Asterisk crashes when queueing video with format | (Reported by Michael Welk) | |
| Remote URL in playback must end with file extension | (Reported by Caesar) | |
| STUN timeout is silently delaying calls | (Reported by Sébastien Duthil) | |
| ari: Audiosocket segfault when no data specified | (Reported by Igor Goncharovsky) | |
| Updated identify/match syntax not supported by config wizard | (Reported by Sean Bright) | |
| fixedjitterbuffer contains an un-wrappered assert that triggers on a negative time slew | (Reported by Dan Cropp) | |
| core: Inband generation of tones for Busy() and Congestion() may not occur | (Reported by Joshua C. Colp) | |
| [patch] Channels are not put on hold for Session Progress with inactive audio | (Reported by Bernd Zobl) | |
| SayNumber triggers WARNING if caller hangs up during application execution | (Reported by N A) | |
| Consolidate res_pjsip_messaging fixes for domain name | (Reported by George Joseph) | |
| Core reload making TCP endpoints go offline | (Reported by Luke Escude) | |
| "FRACK!, Failed assertion bad magic number" happens when unsubscribe an application from an event source | (Reported by Lucas Tardioli Silveira) | |
| Multidomain support issue | (Reported by Andrea Sannucci) | |
| res_rtp_asterisk: Server reflexive candidates use incorrect raddr for RTCP | (Reported by Chris) | |
| pjsip: Asterisk isn't tolerant of RFC8760 UASs | (Reported by George Joseph) | |
| [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY event: dialog XML body | (Reported by Marco Paland) | |
| file.c switch does not account for flash events | (Reported by N A) | |
| chan_sip does not recognize application/hook-flash | (Reported by N A) | |
| cpool_release_pool "double free or corruption (out)" | (Reported by Robert Sutton) | |
| chan_pjsip: Trace message for progress is output even if frame is not queued | (Reported by Michael Maier) | |
| res_rtp_asterisk: Additional RTP-frame (with wrong SSRC) gets inserted when switching from progress to established | (Reported by Matthias Hensler) | |
| chan_local: Filtering audio formats should not occur on removed streams | (Reported by Joshua C. Colp) | |
| translate.c: possible buffer overflow when upsampling | (Reported by Jean Aunis - Prescom) | |
| Segfault - ast_channel_is_multistream (chan=0x0) at channel_internal_api.c:1590 | (Reported by Ross Beer) | |
| prometheus: Crash when scraping bridge | (Reported by Francisco Correia) | |
| res_rtp_asterisk: standard deviation miscalculation | (Reported by Kevin Harwell) | |
| res_rtp_asterisk: Flash events are duplicated | (Reported by N A) | |
| app_queue: CLI set ringinuse for realtime member not working | (Reported by Michael) | |
| Fix differing usage of assignment operators in modules.conf | (Reported by Rusty Newton) | |
| app_queue: updatecdr option in queues.conf does effectively nothing | (Reported by Alexander Gonchiy) | |
| Incorrect description of option "context" in queues.conf.sample | (Reported by Etienne Lessard) | |
| dateformat not read from logger.conf by remote console | (Reported by Igor Liferenko) | |
| app_queue: When "queue show" CLI command is executed a crash occurs | (Reported by Miguel Sanz) | |
| res_pjsip_session: NULL active_media_state topology caused asterisk crash | (Reported by sungtae kim) | |
| app_queue: Queue member status message sent even if status doesn't change | (Reported by Roman Pertsev) | |
| chan_local: Multistream support breaks T.38 faxing | (Reported by Matthias Hensler) | |
| res_pjsip: Allow partial reloading of transports | (Reported by Joshua C. Colp) | |
| menuselect doesn't return errors in many cases | (Reported by George Joseph) | |
| res_rtp_asterisk: Fix frame delivery time when SSRC changes | (Reported by Joshua C. Colp) | |
| app_confbridge: Memory rises when jitterbuffer enabled and muting over AMI occurs | (Reported by Stefan Ruf) | |
| app_dial: DTMF to 'D' option gets duplicated if there are multiple progress events | (Reported by N A) | |
| strings: Incorrect use of __attribute__((pure)) in ast_str_to_lower definition | (Reported by Vitezslav Novy) | |
| res_rtp_asterisk: When native local bridging the remote SSRC becomes permanent | (Reported by Sebastian Damm) | |
| res_pjsip_nat: Contact is rewritten on REGISTER responses with external_signaling_address | (Reported by Brian Paboojian) | |
| ICE Role conflict with an unauthorized session | (Reported by Salah Ahmed) | |
| chan_pjsip: 180 Ringing with SDP not changed into progress | (Reported by Sebastian Damm) | |
| say: Y2021 problem â?? Asterisk cannot say year 2021 in Dutch | (Reported by Jacek Konieczny) | |
| res_pjsip: re-registration gets stuck if setting initial auth credentials fails | (Reported by Nick French) | |
| 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) | |
| Callee declined when 'beep' audio file does not exist | (Reported by IAMJames_) | |
| res_pjsip_refer: Segfault in progress notify | (Reported by George Joseph) | |
| pjsip: | (Reported by Michael Maier) | |
| res_odbc_transaction sets forcecommit default value based on isolation level instead of forcecommit | (Reported by Jaco Kroon) | |
| pjsip: Re-invite occurs when it shouldn't | (Reported by Benjamin Keith Ford) | |
| res_config_pgsql: Limit realtime_pgsql() to return one (no more) record | (Reported by Boris P. Korzun) | |
| app_queue: Member device state "invalid" when second call is ringing and hint is used | (Reported by Boolah ) | |
| app.h: C++ compatibility broken | (Reported by Jean Aunis - Prescom) | |
| res_pjsip_t38: Crash when changing state | (Reported by Gregory Massel) | |
| res_rtp_asterisk: Asterisk crashes when making hold/unhold from webrtc client | (Reported by Edvin Vidmar) | |
| res_pjsip: Segmentation fault | (Reported by Mauri de Souza Meneguzzo (3CPlus)) | |
| chan_sip: Allow peers without audio (text+video). | (Reported by Alexander Traud) | |
| chan_sip: Allow text+video media streams, again. | (Reported by Alexander Traud) | |
| channel: Allow text+video media streams, again. | (Reported by Alexander Traud) | |
| res_pjsip: user=phone validation fail for isup numbers containing *# | (Reported by Mark Petersen) | |
| chan_sip: Audio stream rejected, Other stream present: Invalid SDP. | (Reported by Alexander Traud) | |
| res_pjsip_session: res sometimes uninitialized reported by compiler Clang. | (Reported by Alexander Traud) | |
| 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) | |
| Stasis/messaging: text messages not dispatched to all subscribers when using generic subscription | (Reported by Jean Aunis - Prescom) | |
| chan_sip: SDP: Offers without any enabled stream are accepted. | (Reported by Alexander Traud) | |
| chan_sip: SDP: m=video is parsed even when disabled. | (Reported by Alexander Traud) | |
| chan_sip: Hold/Resume an sRTP call on a video enabled user-agent. | (Reported by Alexander Traud) | |
| chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN instead of a channel variable | (Reported by Ivan Poddubny) | |
| chan_pjsip isn't updating hangupcause on 4XX responses | (Reported by George Joseph) | |
| PJSIP sends duplicate 183 Progress responses | (Reported by Alex Hermann) | |
| chan_pjsip: Subsequent same responses are not stopped | (Reported by Julien) | |
| pjsip: Asterisk goes crazy and massively spams logfile if registration can't be send | (Reported by Michael Maier) | |
| pjsip: SIGSEGV in CLI if no trunk is registered | (Reported by Michael Maier) | |
| LOCK() can grant the same lock to multiple channels spuriously | (Reported by Jaco Kroon) | |
| Segmentation fault in mixmonitor_ds_destroy | (Reported by Robert Sutton) | |
| Crash occurs when Transfer and execute Hangup before the Transfer result | (Reported by Dan Cropp) | |
| Asterisk crashes during call transfer | (Reported by Dalius Mockevicius) | |
| res_pjsip: Crash when examining transport | (Reported by N GM ) | |
| tel: URI in Diversion header causes crash | (Reported by Mikhail Ivanov) | |
| Spyee information ist missing in ChanSpyStop AMI Event | (Reported by Hendrik Wedhorn) | |
| null media causing the Asterisk crash | (Reported by sungtae kim) | |
| Debug messages printed by scope trace might be missing newlines | (Reported by Alexander Traud) | |
| pjsip: Route Header in Cancel request incorrectly set | (Reported by Flole Systems) | |
| res_musiconhold: Segfault on realtime music on hold without entries | (Reported by Nathan Bruning) | |
| Crash when manipulating PJSIP invite dlg ref counts | (Reported by Sean Bright) | |
| Media cache URL requests allow infinite redirects | (Reported by Sean Bright) | |
| res_pjsip_stir_shaken: Fix module description | (Reported by Stanislav Abramenkov) | |
| AST_MODULE_INFO no, MODULEINFO depend | (Reported by Alexander Traud) | |
| res_pjsip: malformed header Accept-Encoding in OPTIONS response | (Reported by Alexander Greiner-Baer) | |
| [patch] chan_sip: TCP/TLS client without server. | (Reported by Alexander Traud) | |
| Incorrect setup of recall channels | (Reported by Boris P. Korzun) | |
| app_queue: Deadlock between queues container and individual queues | (Reported by George Joseph) | |
| res_pjsip.so fails to load when bundled pjproject is compiled without libssl | (Reported by Walter Doekes) | |
| Any curl response checks out as valid even if 404 is returned. | (Reported by dovid) | |
| res_pjsip: Asterisk doesn't stop sending invites (with auth) on 407 replies | (Reported by Sebastian Damm) | |
| sip_to_pjsip.py: doesn't read globbed includes | (Reported by Michael Newton) | |
| GCC Warnings with OPTIMIZE=-Og make | (Reported by Alexander Traud) | |
| GCC Warnings: â??%sâ?? directive argument is null. | (Reported by Alexander Traud) | |
| GCC Warnings with OPTIMIZE=-Os make | (Reported by Alexander Traud) | |
| res_pjsip: flow transport broken for outbound requests | (Reported by Nick French) | |
| config: Sample features.conf incorrectly includes " around sound files | (Reported by Benjamin M.) | |
| logger.conf.sample missing comment mark on line 115 | (Reported by Andrew Siplas) | |
| res_pjsip_session: Asterisk 18 does not progress calls due to codec negotiation after upgrading from Asterisk 16 | (Reported by Ross Beer) | |
| res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF | (Reported by under) | |
| resource_endpoints.c : Memory leak if endpoint not found | (Reported by Jean Aunis - Prescom) | |
| res_pjsip_config_wizard: Crash when freeing string when failing to add extension | (Reported by Vieri) | |
| app_voicemail: Undocumented behavior from VMSayName | (Reported by Eric Smith) | |
| res_pjsip_sdp_rtp: Does not set correct values on RTP instance when "auto" DTMF is used | (Reported by Sebastian Damm) | |
| res_musiconhold: Realtime MOH only loads a single entry | (Reported by laszlovl) | |
| Crash when ast_translator_build_path fails | (Reported by Jasper van der Neut) | |
| dsp: ast_dsp_silence_noise_with_energy wrong judgment of frame format | (Reported by �家建) | |
| func_curl: Segmentation fault when using CURL after setting httpheader CURLOPT | (Reported by Péter Juhász) | |
| Music On Hold announcement cuts intro of music the first time it is played | (Reported by Thomas Frederiksen) | |
| RTP Ports not cleared after hangup | (Reported by Ross Beer) | |
| res_stasis: Add compare function for bridges moh container | (Reported by Hajek Michal) | |
| Unable to get rtp codec payload code for slin | (Reported by Brian J. Murrell) | |
| res_pjsip_session: Re-INVITE collisions aren't handled correctly | (Reported by George Joseph) | |
| 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) | |
| app_voicemail: When a voicemail is marked as "Urgent", it is not sent by email/processed by the mailcmd command | (Reported by Leandro Dardini) | |
| Lastpause of realtime members is reseting | (Reported by Evandro César Arruda) | |
| 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) | |
| [patch] 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) |
Improvements made in this release:
-----------------------------------
| Add support for future dates in Say.c | (Reported by Shloime Rosenblum) | |
| PJSIP remove_existing unavailable contacts | (Reported by Joseph Nadiv) | |
| func_vmcount: Add support for multiple mailboxes | (Reported by N A) | |
| Support of MIME-type for wav16 | (Reported by Boris P. Korzun) | |
| Add custom logging level | (Reported by N A) | |
| res_pjsip: OLI/ANI2 support missing | (Reported by N A) | |
| app_stack: Include calling location if attempting to branch to nonexistent location | (Reported by N A) | |
| Add option to Application_VoiceMail to suppress instructions only when a custom greeting is present | (Reported by Charlie Smurthwaite) | |
| chan_iax2: Add ANI2 | (Reported by N A) | |
| STUN server address refresh | (Reported by Sébastien Duthil) | |
| bridge_basic: Don't throw warning if attended transfer is cancelled | (Reported by N A) | |
| Media Cache - Delayed remote sound file retrieve delays all playbacks | (Reported by Andre Barbosa) | |
| Return integer instead of float if response is a whole number | (Reported by N A) | |
| app_morsecode: Add American Morse code | (Reported by N A) | |
| app_originate: Allow specifying codec(s) to use | (Reported by N A) | |
| Add support for multiple files for agent announcements | (Reported by N A) | |
| res_http_media_cache: Cleanup audio format lookup in HTTP requests | (Reported by Sean Bright) | |
| ARI - Stasis Playback doesn't hangup call when processing a list of invalid files | (Reported by Andre Barbosa) | |
| ARI - PlaybackFinish skip error events | (Reported by Andre Barbosa) | |
| Allow setting channel variables using Originate application | (Reported by N A) | |
| Recognize application/hook-flash in PJSIP | (Reported by N A) | |
| Missing configuration from PJSIP to SIP conversion script | (Reported by N A) | |
| Asterisk reveals pjproject version in STUN packets | (Reported by Jeremy Lainé) | |
| Add Flash AMI event to handle flash events | (Reported by N A) | |
| Silent voicemail option is not completely silent | (Reported by N A) | |
| loader: Let's output warnings for deprecated modules! | (Reported by Joshua C. Colp) | |
| menuselect: Add ability to set deprecated in and removed in versions for modules | (Reported by Joshua C. Colp) | |
| xml: Embed module information into core XML documentation. | (Reported by Joshua C. Colp) | |
| documentation: Fix inconsistent support levels | (Reported by Joshua C. Colp) | |
| sorcery: Add support for more intelligent reloading. | (Reported by Joshua C. Colp) | |
| res_pjsip_registrar: Include source IP address and port in log messages | (Reported by Joshua C. Colp) | |
| asterisk: Update copyright/company | (Reported by Joshua C. Colp) | |
| Add MixMonitorStart / Stop / Mute AMI events | (Reported by Sébastien Duthil) | |
| TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER) failure SIP code | (Reported by Dan Cropp) | |
| Support of various URL-schemes by MoH | (Reported by Boris P. Korzun) | |
| Two repeated 183 | (Reported by Gant Liu) | |
| contrib: systemd asterisk service for centos8 or other newer linux versions | (Reported by Mark Petersen) | |
| res_http_media_cache: HTTP media cache stored hardcoded in /tmp | (Reported by laszlovl) | |
| VoiceMail() should have an option to play greetings as Early Media | (Reported by Juan Carlos Castro y Castro) | |
| Logger: Add debug logging categories | (Reported by Kevin Harwell) | |
| Do not build chan_sip by default as it is now deprecated | (Reported by Sean Bright) | |
| Increase reg_server column size for ps_contacts table realtime | (Reported by sungtae kim) | |
| Create a Bridge with video_single mode | (Reported by sungtae kim) |
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
Thank you for your continued support of Asterisk!
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