The release of Asterisk 20.0.0 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:
-----------------------------------
| res_stir_shaken: Blind SSRF vulnerabilities | (Reported by Clint Ruoho) | |
| res_stir_shaken: Resource exhaustion with large files | (Reported by Benjamin Keith Ford) | |
| ${SQL_ESC()} not correctly escaping a terminating \ | (Reported by Leandro Dardini) | |
| Crash in PJSIP TLS transport | (Reported by Andrew Yager) | |
| chan_pjsip: Remote denial of service by an authenticated user | (Reported by Ivan Poddubny) |
New Features made in this release:
-----------------------------------
| Add test support to calling external processes | (Reported by Philip Prindeville) | |
| locks: add AMI event for deadlock | (Reported by N A) | |
| app_confbridge: Add end_marked_any option | (Reported by N A) | |
| res_pjsip: Add support for reloading TLS certificate and key information | (Reported by Joshua C. Colp) | |
| features: Add advanced transfer initiation options | (Reported by N A) | |
| db: Add AMI action to retrieve all keys beginning with a prefix | (Reported by N A) | |
| chan_dahdi: Add POLARITY function | (Reported by N A) | |
| cli: Add CLI command to execute a dialplan app | (Reported by N A) | |
| pjsip: Get information from 200 OK INVITE reply headers | (Reported by José Lopes) | |
| pbx: Add pbx helper application | (Reported by N A) | |
| app_voicemail: Add option to prevent deletion of messages | (Reported by N A) | |
| res_parking: Add music on hold override option | (Reported by N A) | |
| res_pjsip_outbound_registration: Make max registration delay configurable | (Reported by N A) | |
| app_confbridge: Add CONFBRIDGE_CHANNELS function | (Reported by N A) | |
| Option to allow a user to not hear the join sound on enter but everyone else can | (Reported by Michael Cargile) | |
| func_db: Add a function to return cardinality of keys at prefix | (Reported by N A) | |
| Hint-like extension value lookup function without device state | (Reported by N A) | |
| chan_pjsip: Add ability to send flash events | (Reported by N A) | |
| cli: Add command to evaluate a function | (Reported by N A) | |
| app_queue: Add music on hold option | (Reported by N A) | |
| func_channel: Add LASTCONTEXT and LASTEXTEN fields | (Reported by N A) | |
| ami: Allow events to be globally disabled | (Reported by N A) | |
| cdr: allow disabling CDR by default | (Reported by N A) | |
| ami: Add AMI event for Wink | (Reported by N A) | |
| app_sf: Add full tech-agnostic SF support | (Reported by N A) | |
| app_sendtext: Add ReceiveText application | (Reported by N A) | |
| func_json: Add JSON parsing function | (Reported by N A) | |
| res_tonedetect: Add call progress tone detection | (Reported by N A) | |
| [patch] app_queue Add Login Time and Last Paused Times to Queue Members | (Reported by Jamuel Starkey) | |
| 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) | |
| Function to asynchronously store digits dialed | (Reported by N A) | |
| AGI channel_status failure | (Reported by bbawkon) |
Bugs fixed in this release:
-----------------------------------
| res_crypto and tests: Memory issues and and uninitialized variable error | (Reported by George Joseph) | |
| res_geolocation: ...may be used uninitialized error in geoloc_config.c | (Reported by George Joseph) | |
| Inbound SIP INVITE with Geo Location causing a Segmentation Fault | (Reported by Dan Cropp) | |
| [res_musiconhold] Allows the moh only for the answered call | (Reported by sungtae kim) | |
| pjsip should support tel uri scheme | (Reported by Gergely Dömsödi) | |
| func_frame_trace: Channel masquerade triggers assertion | (Reported by N A) | |
| res_geolocation: GEOLOC_PROFILE isn't returning correct values on incoming channel | (Reported by George Joseph) | |
| chan_pjsip: Endpoint: allow = all is broken. | (Reported by Alexander Traud) | |
| res_tonedetect: fix typo for frametype | (Reported by N A) | |
| alembic: incoming_call_offer_pref and outgoing_call_offer_pref missing in "ps_endpoints" table | (Reported by Daniel Thümen) | |
| testsuite: Add support for Python 3 | (Reported by Joshua C. Colp) | |
| res_geolocation: Refactor for issues found by users | (Reported by George Joseph) | |
| Memory Leak in Confbridge menu | (Reported by Ted G) | |
| ami: FilterList action doesn't exist | (Reported by N A) | |
| ConfbridgeListRooms Event Not Documented | (Reported by Michael Cargile) | |
| app_meetme: MeetmeList AMI event not documented | (Reported by Michael Cargile) | |
| Documentation doesn't include info about "field", a 3rd required parameter. | (Reported by Chris Young) | |
| pbx_variables: ast_str_strlen can be wrong | (Reported by N A) | |
| OSX: bininstall launchd issue on cross-platfrom build | (Reported by Sergey V. Lobanov) | |
| manager: Global disabled event filtered is incomplete | (Reported by N A) | |
| res_pjsip: no contact-status AMI event on register of prune-on-boot contact that uses the same URI as before Asterisk restart | (Reported by Michael Neuhauser) | |
| chan_dahdi, callerid: Caller ID does not honor presentation | (Reported by N A) | |
| Spelling mistake in configs/samples/queues.conf.sample | (Reported by Sam Banks) | |
| build: Git security vulnerability fix is sad with our accessing git as root during "make install" | (Reported by Joshua C. Colp) | |
| res_pjsip, app_confbridge: Video call through ConfBridge with normal endpoints causes infinite loop/crash | (Reported by N A) | |
| Compile failure in res_geolocation/geoloc_eprofile.c when optimization is enabled | (Reported by George Joseph) | |
| cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is unsupported at this time | (Reported by Morvai Szabolcs) | |
| chan_iax2: Optional dependency on openssl/res_crypto is now mandatory | (Reported by Dmitry Melekhov) | |
| test_aeap_transport: transport_connect_fail sporadically causes failure | (Reported by Kevin Harwell) | |
| features: Update automixmon documentation to reflect reality | (Reported by Trevor Peirce) | |
| pbx_lua: Remove compiler warnings | (Reported by Boris P. Korzun) | |
| res_prometheus: Optional load res_pjsip_outbound_registration.so | (Reported by Boris P. Korzun) | |
| app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy | (Reported by N A) | |
| db: Removing nonexistent entries shows "Database entry removed" | (Reported by N A) | |
| cli: Typing \? freezes the CLI permanently with remote console | (Reported by N A) | |
| app_dial: Allow hook flashes to propogate on outbound dials | (Reported by N A) | |
| res_calendar_icalendar: Microsoft online ICS calendars no longer work | (Reported by N A) | |
| say: Abort if channel hangs up during playback | (Reported by N A) | |
| res_pjsip: allow TLS verification of wildcard cert-bearing servers | (Reported by Kevin Harwell) | |
| console: Recent documentation changes for connecting to remote console are inconsistent | (Reported by Matthias Hensler) | |
| Wrong party is disconnected when hook-flashing on 3-way bridge | (Reported by Josh Alberts) | |
| res_pjsip: UPDATE/re-INVITE not sent when "timers=always" is specified in pjsip.conf | (Reported by Ray Crumrine) | |
| DateTime application: wrong inflection for one o'clock in German | (Reported by Christof Efkemann) | |
| pbx: iax2 switch causes crash due to deadlock and assertion | (Reported by N A) | |
| cli: Targeted debug on startup deadlocks and creates unstable system | (Reported by N A) | |
| res_calendar: Asterisk crashes when starting, and will not run | (Reported by N A) | |
| res_pjsip: No video after un-hold with moh_passthrough=yes | (Reported by Maximilian Fridrich) | |
| [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY event: dialog XML body | (Reported by Marco Paland) | |
| menuselect: libxml include fails under Gentoo | (Reported by waltermoeller) | |
| loader: format warnings in dev mode | (Reported by N A) | |
| pjsip: Open Websocket connection is not reused for outgoing requests | (Reported by LA) | |
| res_pjsip_transport_websocket: Registration over websocket returns a rewritten contact | (Reported by Thomas Guebels) | |
| chan_dahdi: Operator control option borks both lines involved on callee disconnect | (Reported by N A) | |
| GCC 12 issues | (Reported by George Joseph) | |
| res_pjsip_session: No video to caller if no camera available | (Reported by Michael Auracher) | |
| res_pjsip_session: No video after early media | (Reported by Michael Auracher) | |
| chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up Dahdi Call On Hold | (Reported by Josh Alberts) | |
| chan_dahdi: adding ring cadences is not idempotent on dahdi restart | (Reported by N A) | |
| chan_iax2: Prevent crashes due to attempted encryption with missing secrets | (Reported by N A) | |
| menuselect: Disabled by default modules that are enabled are always recompiled | (Reported by N A) | |
| app_meetme: Don't erroneously set global variables when channel is NULL | (Reported by N A) | |
| Asterisk's "T" flag is ignored when used with "r" or "R" flags. (documentation bug) | (Reported by Rusty Newton) | |
| Asterisk seems to ignore the "n" parameter for "disable console colorization" | (Reported by Sebastian Gutierrez) | |
| chan_dahdi: Round robin array size is too small for max number of groups | (Reported by N A) | |
| Session timers get removed on UPDATE | (Reported by Mark Petersen) | |
| file.c: seeking to negative file offset is not prevented | (Reported by N A) | |
| Do not change 180 Ringing to 183 Progress even if early_media already enabled | (Reported by Mark Petersen) | |
| chan_sip: SIP route header is missing on UPDATE | (Reported by Mark Petersen) | |
| iostream: Infinite TCP timeout writing data | (Reported by N A) | |
| Incorrect bridging on transfer | (Reported by Yury Kirsanov) | |
| Failed to sign STIR/SHAKEN payload with functionality not enabled | (Reported by Claude Diderich) | |
| res_pjsip: UDP transport does not work when async_operations is greater than 1 | (Reported by Ross Beer) | |
| ast_variable_list_replace_variable uses variable with new keyword | (Reported by Jasper Hafkenscheid) | |
| cdr_adaptive_odbc: does not support DATETIME database columns | (Reported by Gregory Massel) | |
| pjsip / WebRTC: Chrome creating large number of SDP attributes | (Reported by Josh Hogan) | |
| Crash in pjsip_msg_find_hdr_by_name | (Reported by LA) | |
| Segmentation fault in libasteriskpj.so.2 | (Reported by Daniel Bonazzi) | |
| pbx: Only up to 127 includes in a dialplan context (AST_PBX_MAX_STACK - 1) | (Reported by Tzafrir Cohen) | |
| REGRESSION: The build process is requiring xmllint or xmlstarlet ro be installed when it shouldn't | (Reported by George Joseph) | |
| build: Asterisk 18.11.0 doesn't compile when wget isn't available | (Reported by Stefan Ruijsenaars) | |
| chan_iax2: Fix misaligned spacing in iax2 show netstats printout | (Reported by N A) | |
| chan_iax2: "iax2 show registry" shows host for perceived | (Reported by David Herselman) | |
| res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting channel for lack of RTP activity | (Reported by Dmitriy Serov) | |
| res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP activity in one way sessions | (Reported by Boris P. Korzun) | |
| Adjust for 64bit time_t | (Reported by Andre Heider) | |
| RLS: domain part of 'uri' list attribute mismatch with SUBSCRIBE request | (Reported by Alexei Gradinari) | |
| ari: Retrieving stored recording can returns wrong file | (Reported by Arix) | |
| SayNumber can handle '01' to '07', but not '08' or '09' | (Reported by Jim Van Meggelen) | |
| logging messages truncated when using MUSL runtime | (Reported by Philip Prindeville) | |
| agi: Fix xmldoc bug with set music | (Reported by N A) | |
| documentation: AGICommand_set+music documentation arguments displayed incorreclty | (Reported by Jonathan Harris) | |
| res_config_pgsql: omit "unsupported column type 'text'" error | (Reported by Boris P. Korzun) | |
| docs, LICENSE: pbx.digium.com no longer exists | (Reported by N A) | |
| RLS: Batched Notifications stop working | (Reported by Alexei Gradinari) | |
| taskprocessor: Can cause assert at shutdown | (Reported by Joshua C. Colp) | |
| [patch] Queue Realtime load | (Reported by Alexei Gradinari) | |
| [patch] Realtime queue agents unavailable via AMI before a call event. | (Reported by kwk) | |
| AMI Queuestatus not working (with realtime queue) | (Reported by cagdas kopuz) | |
| res_prometheus: Failure to load causes FRACKs | (Reported by Mark Petersen) | |
| Asterisk AMI sends not-valid XML | (Reported by Napadailo Yaroslav) | |
| res_pjsip_outbound_authenticator_digest: ABRT attempting to clean up auth_sess | (Reported by George Joseph) | |
| res_tonedetect: fix logic errors in code | (Reported by N A) | |
| func_frame_drop: fix buffer usage typo | (Reported by N A) | |
| rtp sequence number can skip after DTMF under certain bridges | (Reported by Torrey Searle) | |
| gethostbyname_r is misdetected on NetBSD and causes a build failure | (Reported by MichaÅ? Górny) | |
| Segfault if sorcery object_lifetime_maximum and qualify_frequency the same value | (Reported by Alexei Gradinari) | |
| make_version uses GNU-ism that break git-svn-id parsing on NetBSD | (Reported by MichaÅ? Górny) | |
| ast_get_tid() not implemented for NetBSD | (Reported by MichaÅ? Górny) | |
| rdtsc is not enabled (stubbed out) on NetBSD | (Reported by MichaÅ? Górny) | |
| Build failure on NetBSD due to hmac function collision | (Reported by MichaÅ? Górny) | |
| res_rtp_asterisk: Invalid comparison creates unreachable code | (Reported by N A) | |
| configure fails if libsrtp dev files are not installed | (Reported by Sean Bright) | |
| res_pjsip_session doesn't support multipart message bodies | (Reported by George Joseph) | |
| Regression: Using external pjproject not working after "hack" commit | (Reported by George Joseph) | |
| VoiceMailMain() fails when encountering non-numeric CALLERID(num) | (Reported by Mark Murawski) | |
| pbx_variables: ASTSBINDIR is missing | (Reported by N A) | |
| It's hard to make changes to bundled pjproject | (Reported by George Joseph) | |
| SAY.CONF wrong logic when converting 24hour time to say 12 hour am/pm | (Reported by Vincent Dubois) | |
| PJSIP processing token with % incorrectly | (Reported by Dan Cropp) | |
| Support for Nordic language syntax in Queues | (Reported by Mark Petersen) | |
| app_queue: QueueSummary and QueueStatus events don't exist in documentation | (Reported by Luke Escude) | |
| tcptls.c: TCP client connect fails due to interrupt | (Reported by Kevin Harwell) | |
| app_queue: extension state incorrect | (Reported by Steve Davies) | |
| SAY_DTMF_INTERRUPT channel variable is not honored | (Reported by Sean Bright) | |
| The ast_rtp_codecs_payloads functions don't preserve order | (Reported by George Joseph) | |
| res_pjsip_sdp_rtp: Codec preference order of remote is not correct on unhold | (Reported by Ross Beer) | |
| Deadlock in bridge_channel_internal_join() on local channels. | (Reported by Krzysztof Trempala) | |
| test_timezone_watch breaks during DST to ST transition | (Reported by Josh Soref) | |
| bundled_pjproject: sip_inv is missing multipart support in some cases | (Reported by George Joseph) | |
| ast_coredumper does not delete results when requested and a specific output dir is set | (Reported by Frederic Van Espen) | |
| pbx_variables: cp4 variables is used uninitialized | (Reported by N A) | |
| pbx_variables: MSet truncates sets after 24 variables | (Reported by N A) | |
| chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause a coredump | (Reported by Mark Petersen) | |
| xmldoc: Dump invalid to XML DTD: XSLT | (Reported by Alexander Traud) | |
| xmldoc: Dump invalid to XML DTD: ACO Matchfield | (Reported by Alexander Traud) | |
| documentation: Doxygen site is no longer being updated | (Reported by Joshua C. Colp) | |
| [patch] Update Doxygen Configuration for make progdocs | (Reported by Andrew Latham) | |
| res_pjsip_sdp_rtp: Warns on every offered crypto suite | (Reported by Alexander Traud) | |
| res_ari: Channel create and dial may cause "BUG! Must supply a channel name.." error | (Reported by Anil Gupta) | |
| Infinite loop when out of ports and rtpstart value is odd | (Reported by Thomas Guebels) | |
| chan_pjsip: Wrong or missing Q.850 reason in CANCEL | (Reported by Simone Lazzaris) | |
| res: Fix for Doxygen | (Reported by Alexander Traud) | |
| main: Fix for Doxygen | (Reported by Alexander Traud) | |
| progdocs: Hidden code sections with syntax errors. | (Reported by Alexander Traud) | |
| progdocs: Fix grouping for latest Doxygen | (Reported by Alexander Traud) | |
| Crash occurs when 2 realtime sippeers mysql connections are configured and we have a schema warning | (Reported by Mario Ban) | |
| stir/shaken: Requires GNU designator | (Reported by Alexander Traud) | |
| progdocs: doxyref.h outdated | (Reported by Alexander Traud) | |
| xmldoc: Fix for Doxygen | (Reported by Alexander Traud) | |
| Segfault in __ao2_ref if refdebug = yes | (Reported by Alexei Gradinari) | |
| channels: Fix for Doxygen | (Reported by Alexander Traud) | |
| bridging: Infinite loop when both Local channel halves in same bridge | (Reported by Joshua C. Colp) | |
| odbc: Fix for Doxygen | (Reported by Alexander Traud) | |
| parking: Fix for Doxygen | (Reported by Alexander Traud) | |
| res_ari: Fix for Doxygen | (Reported by Alexander Traud) | |
| frame: Fix for Doxygen | (Reported by Alexander Traud) | |
| stasis: Fix for Doxygen | (Reported by Alexander Traud) | |
| app: Fix for Doxygen | (Reported by Alexander Traud) | |
| res_xmpp: Fix for Doxygen | (Reported by Alexander Traud) | |
| channel: Fix for Doxygen | (Reported by Alexander Traud) | |
| chan_iax2: Fix for Doxygen | (Reported by Alexander Traud) | |
| res_pjsip: Fix for Doxygen | (Reported by Alexander Traud) | |
| bridges: Fix for Doxygen | (Reported by Alexander Traud) | |
| addons: Fix for Doxygen. | (Reported by Alexander Traud) | |
| apps: Fix for Doxygen | (Reported by Alexander Traud) | |
| tests: Fix for Doxygen | (Reported by Alexander Traud) | |
| progdocs: Avoid multiple use of section labels | (Reported by Alexander Traud) | |
| progdocs: Use Doxygen \example correctly | (Reported by Alexander Traud) | |
| bridge_channel: Fix for Doxygen | (Reported by Alexander Traud) | |
| progdocs: Avoid name with Doxygen \file | (Reported by Alexander Traud) | |
| app_morsecode: Fix deadlock | (Reported by N A) | |
| app_read: Fix custom terminator functionality regression | (Reported by N A) | |
| res_pjsip_callerid: Fix OLI parsing | (Reported by N A) | |
| BuildSystem: In POSIX sh, == in place of = is undefined. | (Reported by Alexander Traud) | |
| pbx: "dialplan reload" is removing minus symbol from dynamic hints | (Reported by Daniel Zanutti) | |
| sig_analog: Fix truncated buffer copy | (Reported by N A) | |
| VoiceMail does not cancel recording on rerecord hangup | (Reported by N A) | |
| res_snmp: Not build on recent Debian distributions. | (Reported by Alexander Traud) | |
| res_config_sqlite: not removed in makeopts.in | (Reported by Alexander Traud) | |
| stasis: Clang 13 warns about the unused but set variable dispatched. | (Reported by Alexander Traud) | |
| aelparse: GCC 11.2 found two maybe uninitialized | (Reported by Alexander Traud) | |
| GCC 11.2: two stringop-overread | (Reported by Alexander Traud) | |
| Squash compiler issues generated by gcc 11 | (Reported by George Joseph) | |
| Using --with-crypto and --with-ssl fails on a recompile | (Reported by George Joseph) | |
| func_talkdetect's logic is completely broken | (Reported by Moritz Fain) | |
| make install downloads x86_32 variants of external modules on non Intel architectures | (Reported by Corey Farrell) | |
| stun: Not all users provide a dst to ast_stun_request | (Reported by Dennis Haney) | |
| [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) |
Improvements made in this release:
-----------------------------------
| res_pjsip_gelocation: Downgrade some NOTICE scope trace debugs to DEBUG level | (Reported by N A) | |
| extend user_eq_phone behavior to local uri's | (Reported by Michael Bradeen) | |
| Reimplement res/res_crypto.c internals with EVP_PKEY interface to Openssl API's | (Reported by Philip Prindeville) | |
| Add test coverage to res/res_crypto.c functionality | (Reported by Philip Prindeville) | |
| pbx_variables: Use const char for pbx_substitute_variables_helper_full_location | (Reported by N A) | |
| res_geolocation: Allow location parameters to be specified in profiles | (Reported by George Joseph) | |
| res_geolocation: Add option to suppress empty elements | (Reported by George Joseph) | |
| res_geolocation: Add built-in profiles to use in fully dynamic configurations | (Reported by George Joseph) | |
| [patch] update RLS to reflect the changes to the lists | (Reported by Alexei Gradinari) | |
| general: fix minor formatting issues | (Reported by N A) | |
| chan_iax2: Add missing option documentation | (Reported by N A) | |
| logger: Improve log levels | (Reported by N A) | |
| cdr.conf: Remove obsolete app_mysql reference | (Reported by N A) | |
| general: Remove obsolete SVN references | (Reported by N A) | |
| Create PJSIP interface module for Geolocation | (Reported by George Joseph) | |
| Create core Geolocation capability for Asterisk | (Reported by George Joseph) | |
| general: fix typos | (Reported by N A) | |
| Upgrade Asterisk to bundled pjproject 2.12.1 | (Reported by Stanislav Abramenkov) | |
| xmldocs: Use example tags for examples | (Reported by N A) | |
| [patch] provide a display name for RLS subscriptions | (Reported by Alexei Gradinari) | |
| res_parking: Warn when invalid parking space requested | (Reported by N A) | |
| Evaluate dialplan functions and variables in agi exec | (Reported by Shloime Rosenblum) | |
| ari: expose channel driver's unique id (i.e. Call-ID for chan_sip/chan_pjsip) in ARI channel resource | (Reported by Moritz Fain) | |
| res_pjsip_outbound_registration: Show time remaining until registration lapses | (Reported by N A) | |
| Missing documentation for chan_dahdi dial string ring cadences | (Reported by Scott Griepentrog) | |
| general: Add since tags to xmldocs | (Reported by N A) | |
| samples: Remove obsolete config files | (Reported by N A) | |
| Add Asterisk External Application Protocol (AEAP) implementation | (Reported by Kevin Harwell) | |
| app_mf, app_sf: Return -1 on hangup | (Reported by N A) | |
| app_meetme: Emit warning if conference not found | (Reported by N A) | |
| build: Remove leftover build references | (Reported by N A) | |
| Qualify pjproject 2.12 for Asterisk | (Reported by George Joseph) | |
| Should Readme include information about install_prereq script? | (Reported by Marcel Wagner) | |
| Use pkg-config to find libxml2 headers and libraries | (Reported by Hugh McMaster) | |
| Documentation: Document explanations and examples for possible values of DIALSTATUS | (Reported by Rusty Newton) | |
| build: External binary modules don't use https | (Reported by INVADE International Ltd.) | |
| pbx_builtins: Add missing documentation | (Reported by N A) | |
| app_queue: Add support for withdrawing a call | (Reported by Kfir Itzhak) | |
| Qualify jansson 2.14 for asterisk | (Reported by George Joseph) | |
| channels: Increase core debug levels for chatty debugs | (Reported by N A) | |
| xmldocs: Add since tag | (Reported by N A) | |
| asterisk.h: add macro for curl user agent | (Reported by N A) | |
| curl, stir_shaken: refactor curl code | (Reported by N A) | |
| app_voicemail: Warn if trying to manage nonexistent mailbox | (Reported by N A) | |
| func_db: Warn about malformed key names | (Reported by N A) | |
| cli: add core dump information to core show settings | (Reported by N A) | |
| documentation: Add default attributes to documentation | (Reported by N A) | |
| app_mp3: Document and warn about https incompatibility | (Reported by N A) | |
| app_mf: Allow reading a maximum number of digits | (Reported by N A) | |
| Enable pickup on channel after having received 183 Progress | (Reported by Mark Petersen) | |
| Queue don't play "thank-you" when here is no hold time announcements | (Reported by Mark Petersen) | |
| res_pjsip_sdp_rtp: Keepalive not supported for video streams | (Reported by Luke Escude) | |
| frame.h: fix CNG documentation typo | (Reported by N A) | |
| documentation: Document special system and channel variables | (Reported by N A) | |
| utils.c: Remove all usages of ast_gethostbyname() | (Reported by Sean Bright) | |
| dsp: Define magic number as macro | (Reported by N A) | |
| cli: add module refresh command | (Reported by N A) | |
| app_mp3: Throw warning if attempting to play a nonexistent stream | (Reported by N A) | |
| Documentation is missing for a few AMI Events - Including CDR and events triggered after the QueueStatus action | (Reported by Dafi Ni) | |
| DIALEDPEERNUMBER not set on destination channel for Queue calls | (Reported by Mark Petersen) | |
| app.c: Throw warnings for nonexistent options | (Reported by N A) | |
| Support for Danish language syntax in VM | (Reported by Mark Petersen) | |
| strings: Fix misusage in comment examples | (Reported by N A) | |
| configs: Minor updates to sample configs | (Reported by N A) | |
| pbx: Add public API for more elegant variable substitution with extensions | (Reported by N A) | |
| Incompatibility with newer spandsp releases (3.0.0+) | (Reported by Dustin Marquess) | |
| documentation: Standardize example syntax | (Reported by N A) | |
| app_voicemail: Refactor email generation functions | (Reported by N A) | |
| Add type for JSON stasis message RTCP Report Received/Sent | (Reported by Boris P. Korzun) | |
| Spelling errors | (Reported by Josh Soref) | |
| chan_iax2: Allow both key and secret to be specified at dial time | (Reported by N A) | |
| Add mix option to Playback application for say and filename | (Reported by Shloime Rosenblum) | |
| 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) |
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
Thank you for your continued support of Asterisk!
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