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.
community and would have not been possible without your participation.
This release candidate is available for immediate download at
https://downloads.asterisk.
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:
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moduleinfo: Add replacement module information (Reported by N A) |
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res_monitor: Disable building by default. (Reported by Joshua C. Colp) |
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muted: Remove deprecated application (Reported by Joshua C. Colp) |
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conf2ael: Remove deprecated application (Reported by Joshua C. Colp) |
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res_config_sqlite: Remove deprecated module (Reported by Joshua C. Colp) |
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chan_vpb: Remove deprecated module (Reported by Joshua C. Colp) |
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chan_misdn: Remove deprecated module (Reported by Joshua C. Colp) |
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chan_nbs: Remove deprecated module (Reported by Joshua C. Colp) |
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chan_phone: Remove deprecated module (Reported by Joshua C. Colp) |
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chan_oss: Remove deprecated module (Reported by Joshua C. Colp) |
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cdr_syslog: Remove deprecated module (Reported by Joshua C. Colp) |
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app_dahdiras: Remove deprecated module (Reported by Joshua C. Colp) |
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app_nbscat: Remove deprecated module (Reported by Joshua C. Colp) |
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app_image: Remove deprecated module (Reported by Joshua C. Colp) |
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app_url: Remove deprecated module (Reported by Joshua C. Colp) |
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app_fax: Remove deprecated module (Reported by Joshua C. Colp) |
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app_ices: Remove deprecated module (Reported by Joshua C. Colp) |
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app_mysql: Remove deprecated module (Reported by Joshua C. Colp) |
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cdr_mysql: Remove deprecated module (Reported by Joshua C. Colp) |
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app_meetme: Deprecated in 19, to be removed in 21 (Reported by Joshua C. Colp) |
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app_osploop: Deprecated in 19, to be removed in 21 (Reported by Joshua C. Colp) |
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chan_alsa: Deprecated in 19, to be removed in 21 (Reported by Joshua C. Colp) |
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chan_mgcp: Deprecated in 19, to be removed in 21 (Reported by Joshua C. Colp) |
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chan_skinny: Deprecated in 19, to be removed in 21 (Reported by Joshua C. Colp) |
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res_pktccops: Deprecated in 19, to be removed in 21 (Reported by Joshua C. Colp) |
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app_macro: Deprecated in 16, to be removed in 21 (Reported by Joshua C. Colp) |
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chan_sip: Deprecated in 17, to be removed in 21 (Reported by Joshua C. Colp) |
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res_monitor: Deprecated in 16, to be removed in 21 (Reported by Joshua C. Colp) |
Security bugs fixed in this release:
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chan_pjsip: Remote denial of service by an authenticated user (Reported by Ivan Poddubny) |
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Crash in PJSIP TLS transport (Reported by Andrew Yager) |
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ASTERISK-29203 / AST-2021-002 — Another scenario is causing a crash (Reported by Gregory Massel) |
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sRTP Replay Protection ignored; even tears down long calls (Reported by Alexander Traud) |
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res_pjsip_diversion: sending multiple 181 responses causes memory corruption and crash (Reported by Ivan Poddubny) |
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res_pjsip_diversion: Crash if Tel URI contains History-Info (Reported by Torrey Searle) |
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pjsip: Crash on call rejection during high load (Reported by Sandro Gauci) |
New Features made in this release:
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Add CHANNEL_EXISTS function (Reported by N A) |
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Add SendMF application (Reported by N A) |
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Add STRBETWEEN function (Reported by N A) |
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Add file and directory functions (Reported by N A) |
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Add SAYFILES function (Reported by N A) |
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Add tone detection module (Reported by N A) |
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Option for Read to be able to accept # (Reported by Sta Retji) |
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Add audio scrambler (Reported by N A) |
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Function to drop frames in the TX or RX directions (Reported by N A) |
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Add PJSIP_HEADERS() and ability to read header by pattern (Reported by Igor Goncharovsky) |
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AGI channel_status failure (Reported by bbawkon) |
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Function to asynchronously store digits dialed (Reported by N A) |
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New application to reload modules (Reported by N A) |
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Add application to wait for condition (Reported by N A) |
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app_dial: Expand A option to allow announcement playback to caller (Reported by N A) |
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app_confbridge: New ConfKick application (Reported by N A) |
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app_confbridge: Allow ConfBridge answer to be suppressed (Reported by N A) |
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Minimum and maximum dialplan functions (Reported by N A) |
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func_volume: Volume function can’t be read (Reported by N A) |
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Chan_pjsip does not support unauthenticated OPTIONS ping (Reported by Ross Beer) |
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Implement support for History-Info (Reported by Torrey Searle) |
Bugs fixed in this release:
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– IAX2 Call Encryption Fails with RSA authentication (Reported by Michael Munger) |
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res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform does not support it (Reported by Matthew Kern) |
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app_read: Fix null pointer crash regression (Reported by N A) |
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res_rtp_asterisk: memory leak (Reported by Jean Aunis – Prescom) |
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ari: Listing bridges fails when dialing bridge exists (Reported by Joshua C. Colp) |
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messaging: AMI MessageSend does not support same parameters as dialplan application (Reported by Brian J. Murrell) |
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app_queue: Custom device state using included hints do not update (Reported by N A) |
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Build failure when disabling PJSIP support (Reported by Guido Falsi) |
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pjproject includes trailing whitespace in sdp format attributes (Reported by George Joseph) |
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MP3Player don’ t work with actual mpg123 versions (Reported by Carlos Oliva) |
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ARI external media channel creation doesn’t set option data (Reported by sungtae kim) |
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test_abstract_jb: frames leak (Reported by Corey Farrell) |
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res_snmp: gcc 11 needs -fPIC to compile correctly (Reported by George Joseph) |
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Asterisk is unable to read extended number format terminfo files (Reported by Sean Bright) |
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dns: Core ast_dns_get_nameservers does not support configured IPv6 servers (Reported by Isaac McDonald) |
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ConfBridge errors on creation conference room (Reported by Alexander Zharov) |
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ARI: external media create doesn’t use body parameter (Reported by sungtae kim) |
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app_agent_pool: XML Doc: unterminated entity reference (Reported by Alexander Traud) |
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Subsequent ‘ael reload’ will cause a lock up (Reported by Mark Murawski) |
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app_queue: Core reload resets queue stats, even when keepstats=yes (Reported by Luke Escude) |
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res_rtp_asterisk: sqrt(.) requires the header math.h. (Reported by Alexander Traud) |
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sig_analog: FCG_CAMA fails to signal ANI spill when using MF signaling (Reported by Sarah Autumn) |
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res_pjproject: Can’t map pjproject log messages to Asterisk TRACE (Reported by George Joseph) |
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app_milliwatt: Milliwatt application doesn’t use the proper timings (Reported by N A) |
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chan_mgcp, resp_pktccops ast_debug support (Reported by Tomas Maldonado) |
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aelparse: include of context with timings fails (Reported by Alexander Traud) |
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Segmentation fault at ast_writestream() when write handler not defined (happens with OGG/Speex) (Reported by Ernani José Camargo Azevedo) |
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cdr_adaptive_odbc: Prevent throwing warnings if CDR filtering is used (Reported by N A) |
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statsd: Remove non-standard metric type Meter (Reported by Rijnhard Hessel) |
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app_voicemail2 became a bit silent, lately (Reported by siggi) |
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G729 audio gets corrupted by Asterisk due to smoother (Reported by under) |
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chan_iax2: Asterisk crashes when queueing video with format (Reported by Michael Welk) |
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Remote URL in playback must end with file extension (Reported by Caesar) |
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STUN timeout is silently delaying calls (Reported by Sébastien Duthil) |
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ari: Audiosocket segfault when no data specified (Reported by Igor Goncharovsky) |
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Updated identify/match syntax not supported by config wizard (Reported by Sean Bright) |
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fixedjitterbuffer contains an un-wrappered assert that triggers on a negative time slew (Reported by Dan Cropp) |
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core: Inband generation of tones for Busy() and Congestion() may not occur (Reported by Joshua C. Colp) |
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Channels are not put on hold for Session Progress with inactive audio (Reported by Bernd Zobl) |
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SayNumber triggers WARNING if caller hangs up during application execution (Reported by N A) |
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Consolidate res_pjsip_messaging fixes for domain name (Reported by George Joseph) |
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Core reload making TCP endpoints go offline (Reported by Luke Escude) |
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“FRACK!, Failed assertion bad magic number” happens when unsubscribe an application from an event source (Reported by Lucas Tardioli Silveira) |
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Multidomain support issue (Reported by Andrea Sannucci) |
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res_rtp_asterisk: Server reflexive candidates use incorrect raddr for RTCP (Reported by Chris) |
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pjsip: Asterisk isn’t tolerant of RFC8760 UASs (Reported by George Joseph) |
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Missing RFC4235 tags and attributes in PJSIP NOTIFY event: dialog XML body (Reported by Marco Paland) |
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file.c switch does not account for flash events (Reported by N A) |
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chan_sip does not recognize application/hook-flash (Reported by N A) |
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cpool_release_pool “double free or corruption (out)” (Reported by Robert Sutton) |
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chan_pjsip: Trace message for progress is output even if frame is not queued (Reported by Michael Maier) |
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res_rtp_asterisk: Additional RTP-frame (with wrong SSRC) gets inserted when switching from progress to established (Reported by Matthias Hensler) |
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chan_local: Filtering audio formats should not occur on removed streams (Reported by Joshua C. Colp) |
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translate.c: possible buffer overflow when upsampling (Reported by Jean Aunis – Prescom) |
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Segfault – ast_channel_is_multistream (chan=0x0) at channel_internal_api.c:1590 (Reported by Ross Beer) |
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prometheus: Crash when scraping bridge (Reported by Francisco Correia) |
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res_rtp_asterisk: standard deviation miscalculation (Reported by Kevin Harwell) |
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res_rtp_asterisk: Flash events are duplicated (Reported by N A) |
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app_queue: CLI set ringinuse for realtime member not working (Reported by Michael) |
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Fix differing usage of assignment operators in modules.conf (Reported by Rusty Newton) |
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app_queue: updatecdr option in queues.conf does effectively nothing (Reported by Alexander Gonchiy) |
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Incorrect description of option “context” in queues.conf.sample (Reported by Etienne Lessard) |
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dateformat not read from logger.conf by remote console (Reported by Igor Liferenko) |
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app_queue: When “queue show” CLI command is executed a crash occurs (Reported by Miguel Sanz) |
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res_pjsip_session: NULL active_media_state topology caused asterisk crash (Reported by sungtae kim) |
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app_queue: Queue member status message sent even if status doesn’t change (Reported by Roman Pertsev) |
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chan_local: Multistream support breaks T.38 faxing (Reported by Matthias Hensler) |
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res_pjsip: Allow partial reloading of transports (Reported by Joshua C. Colp) |
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menuselect doesn’t return errors in many cases (Reported by George Joseph) |
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res_rtp_asterisk: Fix frame delivery time when SSRC changes (Reported by Joshua C. Colp) |
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app_confbridge: Memory rises when jitterbuffer enabled and muting over AMI occurs (Reported by Stefan Ruf) |
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app_dial: DTMF to ‘D’ option gets duplicated if there are multiple progress events (Reported by N A) |
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strings: Incorrect use of __attribute__((pure)) in ast_str_to_lower definition (Reported by Vitezslav Novy) |
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res_rtp_asterisk: When native local bridging the remote SSRC becomes permanent (Reported by Sebastian Damm) |
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res_pjsip_nat: Contact is rewritten on REGISTER responses with external_signaling_address (Reported by Brian Paboojian) |
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ICE Role conflict with an unauthorized session (Reported by Salah Ahmed) |
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chan_pjsip: 180 Ringing with SDP not changed into progress (Reported by Sebastian Damm) |
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say: Y2021 problem – Asterisk cannot say year 2021 in Dutch (Reported by Jacek Konieczny) |
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res_pjsip: re-registration gets stuck if setting initial auth credentials fails (Reported by Nick French) |
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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) |
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Callee declined when ‘beep’ audio file does not exist (Reported by IAMJames_) |
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res_pjsip_refer: Segfault in progress notify (Reported by George Joseph) |
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pjsip: of SDP is not incremented though SDP may be changed on reinvite without SDP offer (Reported by Michael Maier) |
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res_odbc_transaction sets forcecommit default value based on isolation level instead of forcecommit (Reported by Jaco Kroon) |
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pjsip: Re-invite occurs when it shouldn’t (Reported by Benjamin Keith Ford) |
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res_config_pgsql: Limit realtime_pgsql() to return one (no more) record (Reported by Boris P. Korzun) |
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app_queue: Member device state “invalid” when second call is ringing and hint is used (Reported by Boolah ) |
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app.h: C++ compatibility broken (Reported by Jean Aunis – Prescom) |
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res_pjsip_t38: Crash when changing state (Reported by Gregory Massel) |
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res_rtp_asterisk: Asterisk crashes when making hold/unhold from webrtc client (Reported by Edvin Vidmar) |
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res_pjsip: Segmentation fault (Reported by Mauri de Souza Meneguzzo (3CPlus)) |
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chan_sip: Allow peers without audio (text+video). (Reported by Alexander Traud) |
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chan_sip: Allow text+video media streams, again. (Reported by Alexander Traud) |
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channel: Allow text+video media streams, again. (Reported by Alexander Traud) |
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res_pjsip: user=phone validation fail for isup numbers containing *# (Reported by Mark Petersen) |
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chan_sip: Audio stream rejected, Other stream present: Invalid SDP. (Reported by Alexander Traud) |
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res_pjsip_session: res sometimes uninitialized reported by compiler Clang. (Reported by Alexander Traud) |
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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) |
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Stasis/messaging: text messages not dispatched to all subscribers when using generic subscription (Reported by Jean Aunis – Prescom) |
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chan_sip: SDP: Offers without any enabled stream are accepted. (Reported by Alexander Traud) |
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chan_sip: SDP: m=video is parsed even when disabled. (Reported by Alexander Traud) |
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chan_sip: Hold/Resume an sRTP call on a video enabled user-agent. (Reported by Alexander Traud) |
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chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN instead of a channel variable (Reported by Ivan Poddubny) |
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chan_pjsip isn’t updating hangupcause on 4XX responses (Reported by George Joseph) |
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PJSIP sends duplicate 183 Progress responses (Reported by Alex Hermann) |
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chan_pjsip: Subsequent same responses are not stopped (Reported by Julien) |
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pjsip: Asterisk goes crazy and massively spams logfile if registration can’t be send (Reported by Michael Maier) |
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pjsip: SIGSEGV in CLI if no trunk is registered (Reported by Michael Maier) |
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LOCK() can grant the same lock to multiple channels spuriously (Reported by Jaco Kroon) |
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Segmentation fault in mixmonitor_ds_destroy (Reported by Robert Sutton) |
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Crash occurs when Transfer and execute Hangup before the Transfer result (Reported by Dan Cropp) |
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Asterisk crashes during call transfer (Reported by Dalius Mockevicius) |
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res_pjsip: Crash when examining transport (Reported by N GM ) |
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tel: URI in Diversion header causes crash (Reported by Mikhail Ivanov) |
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Spyee information ist missing in ChanSpyStop AMI Event (Reported by Hendrik Wedhorn) |
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null media causing the Asterisk crash (Reported by sungtae kim) |
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Debug messages printed by scope trace might be missing newlines (Reported by Alexander Traud) |
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pjsip: Route Header in Cancel request incorrectly set (Reported by Flole Systems) |
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res_musiconhold: Segfault on realtime music on hold without entries (Reported by Nathan Bruning) |
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Crash when manipulating PJSIP invite dlg ref counts (Reported by Sean Bright) |
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Media cache URL requests allow infinite redirects (Reported by Sean Bright) |
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res_pjsip_stir_shaken: Fix module description (Reported by Stanislav Abramenkov) |
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AST_MODULE_INFO no, MODULEINFO depend (Reported by Alexander Traud) |
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res_pjsip: malformed header Accept-Encoding in OPTIONS response (Reported by Alexander Greiner-Baer) |
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chan_sip: TCP/TLS client without server. (Reported by Alexander Traud) |
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Incorrect setup of recall channels (Reported by Boris P. Korzun) |
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app_queue: Deadlock between queues container and individual queues (Reported by George Joseph) |
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res_pjsip.so fails to load when bundled pjproject is compiled without libssl (Reported by Walter Doekes) |
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Any curl response checks out as valid even if 404 is returned. (Reported by dovid) |
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res_pjsip: Asterisk doesn’t stop sending invites (with auth) on 407 replies (Reported by Sebastian Damm) |
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sip_to_pjsip.py: doesn’t read globbed includes (Reported by Michael Newton) |
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GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander Traud) |
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GCC Warnings: ‘%s’ directive argument is null. (Reported by Alexander Traud) |
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GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander Traud) |
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res_pjsip: flow transport broken for outbound requests (Reported by Nick French) |
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config: Sample features.conf incorrectly includes ” around sound files (Reported by Benjamin M.) |
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logger.conf.sample missing comment mark on line 115 (Reported by Andrew Siplas) |
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res_pjsip_session: Asterisk 18 does not progress calls due to codec negotiation after upgrading from Asterisk 16 (Reported by Ross Beer) |
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res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF (Reported by under) |
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resource_endpoints.c : Memory leak if endpoint not found (Reported by Jean Aunis – Prescom) |
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res_pjsip_config_wizard: Crash when freeing string when failing to add extension (Reported by Vieri) |
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app_voicemail: Undocumented behavior from VMSayName (Reported by Eric Smith) |
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res_pjsip_sdp_rtp: Does not set correct values on RTP instance when “auto” DTMF is used (Reported by Sebastian Damm) |
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res_musiconhold: Realtime MOH only loads a single entry (Reported by laszlovl) |
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Crash when ast_translator_build_path fails (Reported by Jasper van der Neut) |
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dsp: ast_dsp_silence_noise_with_ (Reported by 周家建) |
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func_curl: Segmentation fault when using CURL after setting httpheader CURLOPT (Reported by Péter Juhász) |
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Music On Hold announcement cuts intro of music the first time it is played (Reported by Thomas Frederiksen) |
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RTP Ports not cleared after hangup (Reported by Ross Beer) |
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res_stasis: Add compare function for bridges moh container (Reported by Hajek Michal) |
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Unable to get rtp codec payload code for slin (Reported by Brian J. Murrell) |
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res_pjsip_session: Re-INVITE collisions aren’t handled correctly (Reported by George Joseph) |
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Duplicate logging in queue log for EXITEMPTY events (Reported by Ove Aursand) |
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app_queue: Leave empty sometimes not recorded as abandoned (Reported by Kfir Itzhak) |
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res_parking: Parker UUID is no longer copied (Reported by Misha Vodsedalek) |
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chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported by Joseph Ades) |
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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) |
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res_speech: Assertion on format (Reported by Nickolay V. Shmyrev) |
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chan_pjsip does not process or forward 181 responses (Reported by Torrey Searle) |
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app_voicemail: When a voicemail is marked as “Urgent”, it is not sent by email/processed by the mailcmd command (Reported by Leandro Dardini) |
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Lastpause of realtime members is reseting (Reported by Evandro César Arruda) |
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res_pjsip_session: Aggressively terminates session on failed re-INVITE (Reported by Joshua C. Colp) |
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res_rtp_asterisk: T.140 messages have appended RTP string to each message block. (Reported by Thomas Johnson) |
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chan_sip: ToHost property not cleared on reload (Reported by Dennis) |
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Fix VERSION(ASTERISK_VERSION_NUM) on certified versions (Reported by cmaj) |
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Asterisk crash in music on hold (Reported by David Cunningham) |
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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) |
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res_pjsip_registrar: Expires on statically configured contacts is not correct (Reported by tootai) |
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BridgeCreated ARI event shows wrong video_mode info (Reported by sungtae kim) |
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acl: named_acl rule misconfiguration results in segfault on reading rule from realtime (Reported by Andrew Yager) |
Improvements made in this release:
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Add support for future dates in Say.c (Reported by Shloime Rosenblum) |
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PJSIP remove_existing unavailable contacts (Reported by Joseph Nadiv) |
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func_vmcount: Add support for multiple mailboxes (Reported by N A) |
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Support of MIME-type for wav16 (Reported by Boris P. Korzun) |
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Add custom logging level (Reported by N A) |
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res_pjsip: OLI/ANI2 support missing (Reported by N A) |
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app_stack: Include calling location if attempting to branch to nonexistent location (Reported by N A) |
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Add option to Application_VoiceMail to suppress instructions only when a custom greeting is present (Reported by Charlie Smurthwaite) |
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chan_iax2: Add ANI2 (Reported by N A) |
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STUN server address refresh (Reported by Sébastien Duthil) |
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bridge_basic: Don’t throw warning if attended transfer is cancelled (Reported by N A) |
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Media Cache – Delayed remote sound file retrieve delays all playbacks (Reported by Andre Barbosa) |
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Return integer instead of float if response is a whole number (Reported by N A) |
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app_morsecode: Add American Morse code (Reported by N A) |
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app_originate: Allow specifying codec(s) to use (Reported by N A) |
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Add support for multiple files for agent announcements (Reported by N A) |
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res_http_media_cache: Cleanup audio format lookup in HTTP requests (Reported by Sean Bright) |
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ARI – Stasis Playback doesn’t hangup call when processing a list of invalid files (Reported by Andre Barbosa) |
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ARI – PlaybackFinish skip error events (Reported by Andre Barbosa) |
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Allow setting channel variables using Originate application (Reported by N A) |
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Recognize application/hook-flash in PJSIP (Reported by N A) |
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Missing configuration from PJSIP to SIP conversion script (Reported by N A) |
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Asterisk reveals pjproject version in STUN packets (Reported by Jeremy Lainé) |
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Add Flash AMI event to handle flash events (Reported by N A) |
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Silent voicemail option is not completely silent (Reported by N A) |
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loader: Let’s output warnings for deprecated modules! (Reported by Joshua C. Colp) |
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menuselect: Add ability to set deprecated in and removed in versions for modules (Reported by Joshua C. Colp) |
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xml: Embed module information into core XML documentation. (Reported by Joshua C. Colp) |
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documentation: Fix inconsistent support levels (Reported by Joshua C. Colp) |
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sorcery: Add support for more intelligent reloading. (Reported by Joshua C. Colp) |
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res_pjsip_registrar: Include source IP address and port in log messages (Reported by Joshua C. Colp) |
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asterisk: Update copyright/company (Reported by Joshua C. Colp) |
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Add MixMonitorStart / Stop / Mute AMI events (Reported by Sébastien Duthil) |
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TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER) failure SIP code (Reported by Dan Cropp) |
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Support of various URL-schemes by MoH (Reported by Boris P. Korzun) |
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Two repeated 183 (Reported by Gant Liu) |
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contrib: systemd asterisk service for centos8 or other newer linux versions (Reported by Mark Petersen) |
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res_http_media_cache: HTTP media cache stored hardcoded in /tmp (Reported by laszlovl) |
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VoiceMail() should have an option to play greetings as Early Media (Reported by Juan Carlos Castro y Castro) |
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Logger: Add debug logging categories (Reported by Kevin Harwell) |
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Do not build chan_sip by default as it is now deprecated (Reported by Sean Bright) |
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Increase reg_server column size for ps_contacts table realtime (Reported by sungtae kim) |
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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.
Thank you for your continued support of Asterisk!