[asterisk-dev] Asterisk 19.0.0-rc1 Now Available
Asterisk Development Team
asteriskteam at digium.com
Wed Oct 13 06:51:14 CDT 2021
The Asterisk Development Team would like to announce the first
release candidate of Asterisk 19.0.0.
This release candidate is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 19.0.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release candidate:
Deprecations made in this release:
-----------------------------------
* ASTERISK-29601 - moduleinfo: Add replacement module
information
(Reported by N A)
* ASTERISK-29602 - res_monitor: Disable building by default.
(Reported by Joshua C. Colp)
* ASTERISK-29600 - muted: Remove deprecated application
(Reported by Joshua C. Colp)
* ASTERISK-29599 - conf2ael: Remove deprecated application
(Reported by Joshua C. Colp)
* ASTERISK-29598 - res_config_sqlite: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29597 - chan_vpb: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29596 - chan_misdn: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29595 - chan_nbs: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29594 - chan_phone: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29593 - chan_oss: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29592 - cdr_syslog: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29591 - app_dahdiras: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29590 - app_nbscat: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29589 - app_image: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29588 - app_url: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29587 - app_fax: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29586 - app_ices: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29585 - app_mysql: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29584 - cdr_mysql: Remove deprecated module
(Reported by Joshua C. Colp)
* ASTERISK-29548 - app_meetme: Deprecated in 19, to be removed
in 21
(Reported by Joshua C. Colp)
* ASTERISK-29549 - app_osploop: Deprecated in 19, to be removed
in 21
(Reported by Joshua C. Colp)
* ASTERISK-29550 - chan_alsa: Deprecated in 19, to be removed
in 21
(Reported by Joshua C. Colp)
* ASTERISK-29551 - chan_mgcp: Deprecated in 19, to be removed
in 21
(Reported by Joshua C. Colp)
* ASTERISK-29552 - chan_skinny: Deprecated in 19, to be removed
in 21
(Reported by Joshua C. Colp)
* ASTERISK-29553 - res_pktccops: Deprecated in 19, to be
removed in 21
(Reported by Joshua C. Colp)
* ASTERISK-29558 - app_macro: Deprecated in 16, to be removed
in 21
(Reported by Joshua C. Colp)
* ASTERISK-29567 - chan_sip: Deprecated in 17, to be removed in
21
(Reported by Joshua C. Colp)
* ASTERISK-29572 - res_monitor: Deprecated in 16, to be removed
in 21
(Reported by Joshua C. Colp)
Security bugs fixed in this release:
-----------------------------------
* ASTERISK-29381 - chan_pjsip: Remote denial of service by an
authenticated user
(Reported by Ivan Poddubny)
* ASTERISK-29415 - Crash in PJSIP TLS transport
(Reported by Andrew Yager)
* ASTERISK-29305 - ASTERISK-29203 / AST-2021-002 -- Another
scenario is causing a crash
(Reported by Gregory Massel)
* ASTERISK-29260 - sRTP Replay Protection ignored; even tears
down long calls
(Reported by Alexander Traud)
* ASTERISK-29227 - res_pjsip_diversion: sending multiple 181
responses causes memory corruption and crash
(Reported by
Ivan Poddubny)
* ASTERISK-29219 - res_pjsip_diversion: Crash if Tel URI
contains History-Info
(Reported by Torrey Searle)
* ASTERISK-29057 - pjsip: Crash on call rejection during high
load
(Reported by Sandro Gauci)
New Features made in this release:
-----------------------------------
* ASTERISK-29656 - Add CHANNEL_EXISTS function
(Reported
by N A)
* ASTERISK-29496 - Add SendMF application
(Reported by N
A)
* ASTERISK-29627 - Add STRBETWEEN function
(Reported by N
A)
* ASTERISK-29628 - Add file and directory functions
(Reported by N A)
* ASTERISK-29531 - Add SAYFILES function
(Reported by N
A)
* ASTERISK-29546 - Add tone detection module
(Reported by
N A)
* ASTERISK-18454 - Option for Read to be able to accept #
(Reported by Sta Retji)
* ASTERISK-29542 - Add audio scrambler
(Reported by N A)
* ASTERISK-29478 - Function to drop frames in the TX or RX
directions
(Reported by N A)
* ASTERISK-29389 - Add PJSIP_HEADERS() and ability to read
header by pattern
(Reported by Igor Goncharovsky)
* ASTERISK-11 - AGI channel_status failure
(Reported by
bbawkon)
* ASTERISK-29477 - Function to asynchronously store digits
dialed
(Reported by N A)
* ASTERISK-29454 - New application to reload modules
(Reported by N A)
* ASTERISK-29444 - Add application to wait for condition
(Reported by N A)
* ASTERISK-29442 - app_dial: Expand A option to allow
announcement playback to caller
(Reported by N A)
* ASTERISK-29446 - app_confbridge: New ConfKick application
(Reported by N A)
* ASTERISK-29440 - app_confbridge: Allow ConfBridge answer to
be suppressed
(Reported by N A)
* ASTERISK-29431 - Minimum and maximum dialplan functions
(Reported by N A)
* ASTERISK-29439 - func_volume: Volume function can't be read
(Reported by N A)
* ASTERISK-27477 - Chan_pjsip does not support unauthenticated
OPTIONS ping
(Reported by Ross Beer)
* ASTERISK-29027 - Implement support for History-Info
(Reported by Torrey Searle)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-20219 - [patch] - IAX2 Call Encryption Fails with
RSA authentication
(Reported by Michael Munger)
* ASTERISK-29402 - res_pjsip_t38: Socket is bound to IPv4/IPv6
but platform does not support it
(Reported by Matthew
Kern)
* ASTERISK-29673 - app_read: Fix null pointer crash regression
(Reported by N A)
* ASTERISK-29671 - res_rtp_asterisk: memory leak
(Reported by Jean Aunis - Prescom)
* ASTERISK-29668 - ari: Listing bridges fails when dialing
bridge exists
(Reported by Joshua C. Colp)
* ASTERISK-29663 - messaging: AMI MessageSend does not support
same parameters as dialplan application
(Reported by Brian
J. Murrell)
* ASTERISK-29578 - app_queue: Custom device state using
included hints do not update
(Reported by N A)
* ASTERISK-29660 - Build failure when disabling PJSIP support
(Reported by Guido Falsi)
* ASTERISK-29654 - pjproject includes trailing whitespace in
sdp format attributes
(Reported by George Joseph)
* ASTERISK-29635 - MP3Player don' t work with actual mpg123
versions
(Reported by Carlos Oliva)
* ASTERISK-29629 - ARI external media channel creation doesn't
set option data
(Reported by sungtae kim)
* ASTERISK-27176 - test_abstract_jb: frames leak
(Reported by Corey Farrell)
* ASTERISK-29634 - res_snmp: gcc 11 needs -fPIC to compile
correctly
(Reported by George Joseph)
* ASTERISK-29630 - Asterisk is unable to read extended number
format terminfo files
(Reported by Sean Bright)
* ASTERISK-28004 - dns: Core ast_dns_get_nameservers does not
support configured IPv6 servers
(Reported by Isaac
McDonald)
* ASTERISK-29618 - ConfBridge errors on creation conference
room
(Reported by Alexander Zharov)
* ASTERISK-29622 - ARI: external media create doesn't use body
parameter
(Reported by sungtae kim)
* ASTERISK-29614 - app_agent_pool: XML Doc: unterminated entity
reference
(Reported by Alexander Traud)
* ASTERISK-29609 - Subsequent 'ael reload' will cause a lock
up
(Reported by Mark Murawski)
* ASTERISK-28701 - app_queue: Core reload resets queue stats,
even when keepstats=yes
(Reported by Luke Escude)
* ASTERISK-29616 - res_rtp_asterisk: sqrt(.) requires the
header math.h.
(Reported by Alexander Traud)
* ASTERISK-29518 - sig_analog: FCG_CAMA fails to signal ANI
spill when using MF signaling
(Reported by Sarah Autumn)
* ASTERISK-29582 - res_pjproject: Can't map pjproject log
messages to Asterisk TRACE
(Reported by George Joseph)
* ASTERISK-29575 - app_milliwatt: Milliwatt application doesn't
use the proper timings
(Reported by N A)
* ASTERISK-20339 - chan_mgcp, resp_pktccops ast_debug support
(Reported by Tomas Maldonado)
* ASTERISK-29540 - aelparse: include of context with timings
fails
(Reported by Alexander Traud)
* ASTERISK-29539 - Segmentation fault at ast_writestream() when
write handler not defined (happens with OGG/Speex)
(Reported by Ernani Jos�� Camargo Azevedo)
* ASTERISK-29494 - cdr_adaptive_odbc: Prevent throwing warnings
if CDR filtering is used
(Reported by N A)
* ASTERISK-29513 - statsd: Remove non-standard metric type
Meter
(Reported by Rijnhard Hessel)
* ASTERISK-12 - app_voicemail2 became a bit silent, lately
(Reported by siggi)
* ASTERISK-29526 - G729 audio gets corrupted by Asterisk due to
smoother
(Reported by under)
* ASTERISK-29392 - chan_iax2: Asterisk crashes when queueing
video with format
(Reported by Michael Welk)
* ASTERISK-27871 - Remote URL in playback must end with file
extension
(Reported by Caesar)
* ASTERISK-29507 - STUN timeout is silently delaying calls
(Reported by S��bastien Duthil)
* ASTERISK-29514 - ari: Audiosocket segfault when no data
specified
(Reported by Igor Goncharovsky)
* ASTERISK-29503 - Updated identify/match syntax not supported
by config wizard
(Reported by Sean Bright)
* ASTERISK-29480 - fixedjitterbuffer contains an un-wrappered
assert that triggers on a negative time slew
(Reported by
Dan Cropp)
* ASTERISK-29485 - core: Inband generation of tones for Busy()
and Congestion() may not occur
(Reported by Joshua C.
Colp)
* ASTERISK-29479 - [patch] Channels are not put on hold for
Session Progress with inactive audio
(Reported by Bernd
Zobl)
* ASTERISK-29475 - SayNumber triggers WARNING if caller hangs
up during application execution
(Reported by N A)
* ASTERISK-29404 - Consolidate res_pjsip_messaging fixes for
domain name
(Reported by George Joseph)
* ASTERISK-29441 - Core reload making TCP endpoints go offline
(Reported by Luke Escude)
* ASTERISK-28237 - "FRACK!, Failed assertion bad magic number"
happens when unsubscribe an application from an event source
(Reported by Lucas Tardioli Silveira)
* ASTERISK-28393 - Multidomain support issue
(Reported by
Andrea Sannucci)
* ASTERISK-29433 - res_rtp_asterisk: Server reflexive
candidates use incorrect raddr for RTCP
(Reported by
Chris)
* ASTERISK-29397 - pjsip: Asterisk isn't tolerant of RFC8760
UASs
(Reported by George Joseph)
* ASTERISK-24601 - [patch]Missing RFC4235 tags and attributes
in PJSIP NOTIFY event: dialog XML body
(Reported by Marco
Paland)
* ASTERISK-29372 - file.c switch does not account for flash
events
(Reported by N A)
* ASTERISK-29370 - chan_sip does not recognize
application/hook-flash
(Reported by N A)
* ASTERISK-29377 - cpool_release_pool "double free or
corruption (out)"
(Reported by Robert Sutton)
* ASTERISK-29358 - chan_pjsip: Trace message for progress is
output even if frame is not queued
(Reported by Michael
Maier)
* ASTERISK-29030 - res_rtp_asterisk: Additional RTP-frame (with
wrong SSRC) gets inserted when switching from progress to
established
(Reported by Matthias Hensler)
* ASTERISK-29407 - chan_local: Filtering audio formats should
not occur on removed streams
(Reported by Joshua C. Colp)
* ASTERISK-29328 - translate.c: possible buffer overflow when
upsampling
(Reported by Jean Aunis - Prescom)
* ASTERISK-29379 - Segfault - ast_channel_is_multistream
(chan=0x0) at channel_internal_api.c:1590
(Reported by
Ross Beer)
* ASTERISK-29130 - prometheus: Crash when scraping bridge
(Reported by Francisco Correia)
* ASTERISK-29364 - res_rtp_asterisk: standard deviation
miscalculation
(Reported by Kevin Harwell)
* ASTERISK-29373 - res_rtp_asterisk: Flash events are
duplicated
(Reported by N A)
* ASTERISK-28356 - app_queue: CLI set ringinuse for realtime
member not working
(Reported by Michael)
* ASTERISK-24434 - Fix differing usage of assignment operators
in modules.conf
(Reported by Rusty Newton)
* ASTERISK-26614 - app_queue: updatecdr option in queues.conf
does effectively nothing
(Reported by Alexander Gonchiy)
* ASTERISK-24631 - Incorrect description of option "context" in
queues.conf.sample
(Reported by Etienne Lessard)
* ASTERISK-25358 - dateformat not read from logger.conf by
remote console
(Reported by Igor Liferenko)
* ASTERISK-27542 - app_queue: When "queue show" CLI command is
executed a crash occurs
(Reported by Miguel Sanz)
* ASTERISK-29215 - res_pjsip_session: NULL active_media_state
topology caused asterisk crash
(Reported by sungtae kim)
* ASTERISK-29355 - app_queue: Queue member status message sent
even if status doesn't change
(Reported by Roman Pertsev)
* ASTERISK-29035 - chan_local: Multistream support breaks T.38
faxing
(Reported by Matthias Hensler)
* ASTERISK-29354 - res_pjsip: Allow partial reloading of
transports
(Reported by Joshua C. Colp)
* ASTERISK-29348 - menuselect doesn't return errors in many
cases
(Reported by George Joseph)
* ASTERISK-29352 - res_rtp_asterisk: Fix frame delivery time
when SSRC changes
(Reported by Joshua C. Colp)
* ASTERISK-29071 - app_confbridge: Memory rises when
jitterbuffer enabled and muting over AMI occurs
(Reported
by Stefan Ruf)
* ASTERISK-29329 - app_dial: DTMF to 'D' option gets duplicated
if there are multiple progress events
(Reported by N A)
* ASTERISK-29306 - strings: Incorrect use of
__attribute__((pure)) in ast_str_to_lower definition
(Reported by Vitezslav Novy)
* ASTERISK-29300 - res_rtp_asterisk: When native local bridging
the remote SSRC becomes permanent
(Reported by Sebastian
Damm)
* ASTERISK-29235 - res_pjsip_nat: Contact is rewritten on
REGISTER responses with external_signaling_address
(Reported by Brian Paboojian)
* ASTERISK-29266 - ICE Role conflict with an unauthorized
session
(Reported by Salah Ahmed)
* ASTERISK-29105 - chan_pjsip: 180 Ringing with SDP not changed
into progress
(Reported by Sebastian Damm)
* ASTERISK-29297 - say: Y2021 problem ��� Asterisk cannot say
year 2021 in Dutch
(Reported by Jacek Konieczny)
* ASTERISK-29315 - res_pjsip: re-registration gets stuck if
setting initial auth credentials fails
(Reported by Nick
French)
* ASTERISK-29312 - res_fax: asterisk fails to publish the
Stasis and ReceiveFax status messages if the remote Station ID
contains invalid UTF-8 characters
(Reported by Alexei
Gradinari)
* ASTERISK-16799 - Callee declined when 'beep' audio file does
not exist
(Reported by IAMJames_)
* ASTERISK-29313 - res_pjsip_refer: Segfault in progress
notify
(Reported by George Joseph)
* ASTERISK-28452 - pjsip: <sess-version> of SDP is not
incremented though SDP may be changed on reinvite without SDP
offer
(Reported by Michael Maier)
* ASTERISK-29311 - res_odbc_transaction sets forcecommit
default value based on isolation level instead of forcecommit
(Reported by Jaco Kroon)
* ASTERISK-29303 - pjsip: Re-invite occurs when it shouldn't
(Reported by Benjamin Keith Ford)
* ASTERISK-29293 - res_config_pgsql: Limit realtime_pgsql() to
return one (no more) record
(Reported by Boris P. Korzun)
* ASTERISK-28369 - app_queue: Member device state "invalid"
when second call is ringing and hint is used
(Reported by
Boolah )
* ASTERISK-29287 - app.h: C++ compatibility broken
(Reported by Jean Aunis - Prescom)
* ASTERISK-29203 - res_pjsip_t38: Crash when changing state
(Reported by Gregory Massel)
* ASTERISK-29205 - res_rtp_asterisk: Asterisk crashes when
making hold/unhold from webrtc client
(Reported by Edvin
Vidmar)
* ASTERISK-29196 - res_pjsip: Segmentation fault
(Reported by Mauri de Souza Meneguzzo (3CPlus))
* ASTERISK-29280 - chan_sip: Allow peers without audio
(text+video).
(Reported by Alexander Traud)
* ASTERISK-29265 - chan_sip: Allow text+video media streams,
again.
(Reported by Alexander Traud)
* ASTERISK-29259 - channel: Allow text+video media streams,
again.
(Reported by Alexander Traud)
* ASTERISK-29261 - res_pjsip: user=phone validation fail for
isup numbers containing *#
(Reported by Mark Petersen)
* ASTERISK-29258 - chan_sip: Audio stream rejected, Other
stream present: Invalid SDP.
(Reported by Alexander Traud)
* ASTERISK-29248 - res_pjsip_session: res sometimes
uninitialized reported by compiler Clang.
(Reported by
Alexander Traud)
* ASTERISK-29220 - After T38 reinvite response of 488 a
subsequent G711 reinvite is not processed correctly. Instead the
previous T38 session media is used
(Reported by Robert
Cripps)
* ASTERISK-29229 - Stasis/messaging: text messages not
dispatched to all subscribers when using generic subscription
(Reported by Jean Aunis - Prescom)
* ASTERISK-29238 - chan_sip: SDP: Offers without any enabled
stream are accepted.
(Reported by Alexander Traud)
* ASTERISK-29237 - chan_sip: SDP: m=video is parsed even when
disabled.
(Reported by Alexander Traud)
* ASTERISK-29222 - chan_sip: Hold/Resume an sRTP call on a
video enabled user-agent.
(Reported by Alexander Traud)
* ASTERISK-29240 - chan_pjsip: Incoming PJSIP calls set global
SIPDOMAIN instead of a channel variable
(Reported by Ivan
Poddubny)
* ASTERISK-27902 - chan_pjsip isn't updating hangupcause on 4XX
responses
(Reported by George Joseph)
* ASTERISK-28016 - PJSIP sends duplicate 183 Progress
responses
(Reported by Alex Hermann)
* ASTERISK-28185 - chan_pjsip: Subsequent same responses are
not stopped
(Reported by Julien)
* ASTERISK-29230 - pjsip: Asterisk goes crazy and massively
spams logfile if registration can't be send
(Reported by
Michael Maier)
* ASTERISK-29231 - pjsip: SIGSEGV in CLI if no trunk is
registered
(Reported by Michael Maier)
* ASTERISK-29217 - LOCK() can grant the same lock to multiple
channels spuriously
(Reported by Jaco Kroon)
* ASTERISK-28947 - Segmentation fault in mixmonitor_ds_destroy
(Reported by Robert Sutton)
* ASTERISK-29201 - Crash occurs when Transfer and execute
Hangup before the Transfer result
(Reported by Dan Cropp)
* ASTERISK-29168 - Asterisk crashes during call transfer
(Reported by Dalius Mockevicius)
* ASTERISK-29210 - res_pjsip: Crash when examining transport
(Reported by N GM )
* ASTERISK-29191 - tel: URI in Diversion header causes crash
(Reported by Mikhail Ivanov)
* ASTERISK-28883 - Spyee information ist missing in ChanSpyStop
AMI Event
(Reported by Hendrik Wedhorn)
* ASTERISK-29188 - null media causing the Asterisk crash
(Reported by sungtae kim)
* ASTERISK-29209 - Debug messages printed by scope trace might
be missing newlines
(Reported by Alexander Traud)
* ASTERISK-29024 - pjsip: Route Header in Cancel request
incorrectly set
(Reported by Flole Systems)
* ASTERISK-29211 - res_musiconhold: Segfault on realtime music
on hold without entries
(Reported by Nathan Bruning)
* ASTERISK-29022 - Crash when manipulating PJSIP invite dlg ref
counts
(Reported by Sean Bright)
* ASTERISK-29173 - Media cache URL requests allow infinite
redirects
(Reported by Sean Bright)
* ASTERISK-29175 - res_pjsip_stir_shaken: Fix module
description
(Reported by Stanislav Abramenkov)
* ASTERISK-29148 - AST_MODULE_INFO no, MODULEINFO depend
(Reported by Alexander Traud)
* ASTERISK-29165 - res_pjsip: malformed header Accept-Encoding
in OPTIONS response
(Reported by Alexander Greiner-Baer)
* ASTERISK-28798 - [patch] chan_sip: TCP/TLS client without
server.
(Reported by Alexander Traud)
* ASTERISK-29161 - Incorrect setup of recall channels
(Reported by Boris P. Korzun)
* ASTERISK-29155 - app_queue: Deadlock between queues container
and individual queues
(Reported by George Joseph)
* ASTERISK-28933 - res_pjsip.so fails to load when bundled
pjproject is compiled without libssl
(Reported by Walter
Doekes)
* ASTERISK-28825 - Any curl response checks out as valid even
if 404 is returned.
(Reported by dovid)
* ASTERISK-29013 - res_pjsip: Asterisk doesn't stop sending
invites (with auth) on 407 replies
(Reported by Sebastian
Damm)
* ASTERISK-29142 - sip_to_pjsip.py: doesn't read globbed
includes
(Reported by Michael Newton)
* ASTERISK-29144 - GCC Warnings with OPTIMIZE=-Og make
(Reported by Alexander Traud)
* ASTERISK-29146 - GCC Warnings: ���%s��� directive argument is
null.
(Reported by Alexander Traud)
* ASTERISK-29145 - GCC Warnings with OPTIMIZE=-Os make
(Reported by Alexander Traud)
* ASTERISK-29124 - res_pjsip: flow transport broken for
outbound requests
(Reported by Nick French)
* ASTERISK-29136 - config: Sample features.conf incorrectly
includes " around sound files
(Reported by Benjamin M.)
* ASTERISK-29123 - logger.conf.sample missing comment mark on
line 115
(Reported by Andrew Siplas)
* ASTERISK-29109 - res_pjsip_session: Asterisk 18 does not
progress calls due to codec negotiation after upgrading from
Asterisk 16
(Reported by Ross Beer)
* ASTERISK-28430 - res_rtp_asterisk.c: FRACK!, Failed assertion
errno != EBADF
(Reported by under)
* ASTERISK-29108 - resource_endpoints.c : Memory leak if
endpoint not found
(Reported by Jean Aunis - Prescom)
* ASTERISK-29097 - res_pjsip_config_wizard: Crash when freeing
string when failing to add extension
(Reported by Vieri)
* ASTERISK-26424 - app_voicemail: Undocumented behavior from
VMSayName
(Reported by Eric Smith)
* ASTERISK-29051 - res_pjsip_sdp_rtp: Does not set correct
values on RTP instance when "auto" DTMF is used
(Reported
by Sebastian Damm)
* ASTERISK-29099 - res_musiconhold: Realtime MOH only loads a
single entry
(Reported by laszlovl)
* ASTERISK-29091 - Crash when ast_translator_build_path fails
(Reported by Jasper van der Neut)
* ASTERISK-28311 - dsp: ast_dsp_silence_noise_with_energy wrong
judgment of frame format
(Reported by ���������)
* ASTERISK-29085 - func_curl: Segmentation fault when using
CURL after setting httpheader CURLOPT
(Reported by P��ter
Juh��sz)
* ASTERISK-24329 - Music On Hold announcement cuts intro of
music the first time it is played
(Reported by Thomas
Frederiksen)
* ASTERISK-29089 - RTP Ports not cleared after hangup
(Reported by Ross Beer)
* ASTERISK-29081 - res_stasis: Add compare function for bridges
moh container
(Reported by Hajek Michal)
* ASTERISK-28416 - Unable to get rtp codec payload code for
slin
(Reported by Brian J. Murrell)
* ASTERISK-29014 - res_pjsip_session: Re-INVITE collisions
aren't handled correctly
(Reported by George Joseph)
* ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY
events
(Reported by Ove Aursand)
* ASTERISK-29043 - app_queue: Leave empty sometimes not
recorded as abandoned
(Reported by Kfir Itzhak)
* ASTERISK-29042 - res_parking: Parker UUID is no longer
copied
(Reported by Misha Vodsedalek)
* ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken
asterisk 16
(Reported by Joseph Ades)
* ASTERISK-29046 - pbx: Deadlock when doing a reload, while
simultaneously doing an ExtensionState on a pattern match hint
that ends up adding an extension
(Reported by Ramarajan)
* ASTERISK-29040 - res_speech: Assertion on format
(Reported by Nickolay V. Shmyrev)
* ASTERISK-29001 - chan_pjsip does not process or forward 181
responses
(Reported by Torrey Searle)
* ASTERISK-27273 - app_voicemail: When a voicemail is marked as
"Urgent", it is not sent by email/processed by the mailcmd
command
(Reported by Leandro Dardini)
* ASTERISK-29034 - Lastpause of realtime members is reseting
(Reported by Evandro C��sar Arruda)
* ASTERISK-29033 - res_pjsip_session: Aggressively terminates
session on failed re-INVITE
(Reported by Joshua C. Colp)
* ASTERISK-28974 - res_rtp_asterisk: T.140 messages have
appended RTP string to each message block.
(Reported by
Thomas Johnson)
* ASTERISK-29011 - chan_sip: ToHost property not cleared on
reload
(Reported by Dennis)
* ASTERISK-29021 - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on
certified versions
(Reported by cmaj)
* ASTERISK-28927 - Asterisk crash in music on hold
(Reported by David Cunningham)
* ASTERISK-28973 - Malformed IP address in SDP of 2nd SIP timer
triggered INVITE when NAT is active (UDP transport with
external_media_address)
(Reported by Michael Neuhauser)
* ASTERISK-28995 - res_pjsip_registrar: Expires on statically
configured contacts is not correct
(Reported by tootai)
* ASTERISK-28987 - BridgeCreated ARI event shows wrong
video_mode info
(Reported by sungtae kim)
* ASTERISK-28978 - acl: named_acl rule misconfiguration results
in segfault on reading rule from realtime
(Reported by
Andrew Yager)
Improvements made in this release:
-----------------------------------
* ASTERISK-29637 - Add support for future dates in Say.c
(Reported by Shloime Rosenblum)
* ASTERISK-29525 - PJSIP remove_existing unavailable contacts
(Reported by Joseph Nadiv)
* ASTERISK-29661 - func_vmcount: Add support for multiple
mailboxes
(Reported by N A)
* ASTERISK-29275 - Support of MIME-type for wav16
(Reported by Boris P. Korzun)
* ASTERISK-29529 - Add custom logging level
(Reported by
N A)
* ASTERISK-29472 - res_pjsip: OLI/ANI2 support missing
(Reported by N A)
* ASTERISK-29626 - app_stack: Include calling location if
attempting to branch to nonexistent location
(Reported by
N A)
* ASTERISK-29632 - Add option to Application_VoiceMail to
suppress instructions only when a custom greeting is present
(Reported by Charlie Smurthwaite)
* ASTERISK-29605 - chan_iax2: Add ANI2
(Reported by N A)
* ASTERISK-29508 - STUN server address refresh
(Reported
by S��bastien Duthil)
* ASTERISK-29612 - bridge_basic: Don't throw warning if
attended transfer is cancelled
(Reported by N A)
* ASTERISK-29544 - Media Cache - Delayed remote sound file
retrieve delays all playbacks
(Reported by Andre Barbosa)
* ASTERISK-29495 - Return integer instead of float if response
is a whole number
(Reported by N A)
* ASTERISK-29541 - app_morsecode: Add American Morse code
(Reported by N A)
* ASTERISK-29543 - app_originate: Allow specifying codec(s) to
use
(Reported by N A)
* ASTERISK-29528 - Add support for multiple files for agent
announcements
(Reported by N A)
* ASTERISK-29527 - res_http_media_cache: Cleanup audio format
lookup in HTTP requests
(Reported by Sean Bright)
* ASTERISK-29501 - ARI - Stasis Playback doesn't hangup call
when processing a list of invalid files
(Reported by Andre
Barbosa)
* ASTERISK-29464 - ARI - PlaybackFinish skip error events
(Reported by Andre Barbosa)
* ASTERISK-29450 - Allow setting channel variables using
Originate application
(Reported by N A)
* ASTERISK-29460 - Recognize application/hook-flash in PJSIP
(Reported by N A)
* ASTERISK-29459 - Missing configuration from PJSIP to SIP
conversion script
(Reported by N A)
* ASTERISK-29434 - Asterisk reveals pjproject version in STUN
packets
(Reported by Jeremy Lain��)
* ASTERISK-29380 - Add Flash AMI event to handle flash events
(Reported by N A)
* ASTERISK-29349 - Silent voicemail option is not completely
silent
(Reported by N A)
* ASTERISK-29339 - loader: Let's output warnings for deprecated
modules!
(Reported by Joshua C. Colp)
* ASTERISK-29337 - menuselect: Add ability to set deprecated in
and removed in versions for modules
(Reported by Joshua C.
Colp)
* ASTERISK-29335 - xml: Embed module information into core XML
documentation.
(Reported by Joshua C. Colp)
* ASTERISK-29336 - documentation: Fix inconsistent support
levels
(Reported by Joshua C. Colp)
* ASTERISK-29321 - sorcery: Add support for more intelligent
reloading.
(Reported by Joshua C. Colp)
* ASTERISK-29325 - res_pjsip_registrar: Include source IP
address and port in log messages
(Reported by Joshua C.
Colp)
* ASTERISK-29326 - asterisk: Update copyright/company
(Reported by Joshua C. Colp)
* ASTERISK-29244 - Add MixMonitorStart / Stop / Mute AMI
events
(Reported by S��bastien Duthil)
* ASTERISK-29252 - TRANSFERSTATUSPROTOCOL variable to report
Transfer (REFER) failure SIP code
(Reported by Dan Cropp)
* ASTERISK-29262 - Support of various URL-schemes by MoH
(Reported by Boris P. Korzun)
* ASTERISK-28549 - Two repeated 183
(Reported by Gant
Liu)
* ASTERISK-29216 - contrib: systemd asterisk service for
centos8 or other newer linux versions
(Reported by Mark
Petersen)
* ASTERISK-29143 - res_http_media_cache: HTTP media cache
stored hardcoded in /tmp
(Reported by laszlovl)
* ASTERISK-29118 - VoiceMail() should have an option to play
greetings as Early Media
(Reported by Juan Carlos Castro y
Castro)
* ASTERISK-29054 - Logger: Add debug logging categories
(Reported by Kevin Harwell)
* ASTERISK-29083 - Do not build chan_sip by default as it is
now deprecated
(Reported by Sean Bright)
* ASTERISK-29056 - Increase reg_server column size for
ps_contacts table realtime
(Reported by sungtae kim)
* ASTERISK-29055 - Create a Bridge with video_single mode
(Reported by sungtae kim)
For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0-rc1
Thank you for your continued support of Asterisk!
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