Asterisk Development Team
2022-Apr-28 13:43 UTC
[asterisk-announce] Certified Asterisk 18.9-cert1 Now Available
The Asterisk Development Team would like to announce the release of Certified
Asterisk 18.9-cert1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/certified-asterisk
The release of Certified Asterisk 18.9-cert1 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:
-----------------------------------
* 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-29554 - cdr_mysql: Deprecated in 1.8, to be removed
in 19
(Reported by Joshua C. Colp)
* ASTERISK-29555 - app_mysql: Deprecated in 1.8, to be removed
in 19
(Reported by Joshua C. Colp)
* ASTERISK-29557 - app_ices: Deprecated in 16, to be removed in
19
(Reported by Joshua C. Colp)
* ASTERISK-29558 - app_macro: Deprecated in 16, to be removed
in 21
(Reported by Joshua C. Colp)
* ASTERISK-29559 - app_fax: Deprecated in 16, to be removed in
19
(Reported by Joshua C. Colp)
* ASTERISK-29560 - app_url: Deprecated in 16, to be removed in
19
(Reported by Joshua C. Colp)
* ASTERISK-29561 - app_image: Deprecated in 16, to be removed
in 19
(Reported by Joshua C. Colp)
* ASTERISK-29562 - app_nbscat: Deprecated in 16, to be removed
in 19
(Reported by Joshua C. Colp)
* ASTERISK-29563 - app_dahdiras: Deprecated in 16, to be
removed in 19
(Reported by Joshua C. Colp)
* ASTERISK-29564 - cdr_syslog: Deprecated in 16, to be removed
in 19
(Reported by Joshua C. Colp)
* ASTERISK-29565 - chan_oss: Deprecated in 16, to be removed in
19
(Reported by Joshua C. Colp)
* ASTERISK-29566 - chan_phone: Deprecated in 16, to be removed
in 19
(Reported by Joshua C. Colp)
* ASTERISK-29567 - chan_sip: Deprecated in 17, to be removed in
21
(Reported by Joshua C. Colp)
* ASTERISK-29568 - chan_nbs: Deprecated in 16, to be removed in
19
(Reported by Joshua C. Colp)
* ASTERISK-29569 - chan_misdn: Deprecated in 16, to be removed
in 19
(Reported by Joshua C. Colp)
* ASTERISK-29570 - chan_vpb: Deprecated in 16, to be removed in
19
(Reported by Joshua C. Colp)
* ASTERISK-29571 - res_config_sqlite: Deprecated in 16, to be
removed in 19
(Reported by Joshua C. Colp)
* ASTERISK-29572 - res_monitor: Deprecated in 16, to be removed
in 21
(Reported by Joshua C. Colp)
* ASTERISK-29573 - conf2ael: Deprecated in 16, to be removed in
19
(Reported by Joshua C. Colp)
* ASTERISK-29574 - muted: Deprecated in 16, to be removed in
19
(Reported by Joshua C. Colp)
Security bugs fixed in this release:
-----------------------------------
* ASTERISK-29476 - res_stir_shaken: Blind SSRF vulnerabilities
(Reported by Clint Ruoho)
* ASTERISK-29872 - res_stir_shaken: Resource exhaustion with
large files
(Reported by Benjamin Keith Ford)
* ASTERISK-29838 - ${SQL_ESC()} not correctly escaping a
terminating \
(Reported by Leandro Dardini)
* ASTERISK-29945 - pjproject: Security fixes for things
(Reported by Kevin Harwell)
* ASTERISK-29415 - Crash in PJSIP TLS transport
(Reported by Andrew Yager)
* ASTERISK-29381 - chan_pjsip: Remote denial of service by an
authenticated user
(Reported by Ivan Poddubny)
* 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)
* ASTERISK-28589 - chan_sip: Depending on configuration an
INVITE can alter Addr of a peer
(Reported by Andrey V.
T.)
* ASTERISK-28580 - Bypass SYSTEM write permission in manager
action allows system commands execution
(Reported by Eliel
Sarda��ons)
* ASTERISK-28495 - res_pjsip_t38: 200 OK with SDP answer with
declined stream causes crash
(Reported by Alexei
Gradinari)
* ASTERISK-28447 - res_pjsip_messaging: In-dialog MESSAGE with
no body causes crash
(Reported by Gil Richard)
* ASTERISK-28465 - Broken SDP can cause a segfault in a T.38
reINVITE
(Reported by Francesco Castellano)
* ASTERISK-28260 - Asterisk segfault when rtp negotiation is
wrong or fails
(Reported by Sotiris Ganouris)
* ASTERISK-28127 - Buffer overflow for DNS SRV/NAPTR records
(Reported by Jan Hoffmann)
* ASTERISK-28013 - res_http_websocket: Crash when reading HTTP
Upgrade requests
(Reported by Sean Bright)
New Features made in this release:
-----------------------------------
* ASTERISK-29720 - res_tonedetect: Add call progress tone
detection
(Reported by N A)
* ASTERISK-18069 - [patch] app_queue Add Login Time and Last
Paused Times to Queue Members
(Reported by Jamuel Starkey)
* 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)
* ASTERISK-6863 - [patch] allow Asterisk to set high ToS bits
as non-root on Linux
(Reported by Matt Addison)
* ASTERISK-17491 - CURLOPT() needs a "followlocation" parameter
/ "maxredirs" doesn't do anything
(Reported by candrews)
* ASTERISK-28639 - res_pjsip_endpoint_identifier_ip: Add
ability to match on source port
(Reported by Sean Bright)
* ASTERISK-28614 - app_senddtmf: Allow "receiving" DTMF with
PlayDTMF instead of only "sending"
(Reported by laszlovl)
* ASTERISK-28613 - func_curl: CURLOPT cannot set Content-Type
header
(Reported by Martin Tomec)
* ASTERISK-28533 - func_jitterbuffer: Add support for video
synchronization
(Reported by Joshua C. Colp)
* ASTERISK-17808 - [patch] Unregister a realtime moh class
(Reported by Byron Clark)
* ASTERISK-28489 - Channel variable SIPFROMDOMAIN for
chan_pjsip to setup From header URI domain
(Reported by
Stas Kobzar)
* ASTERISK-28403 - Add native Prometheus support to Asterisk
(Reported by Matt Jordan)
* ASTERISK-28375 - res_pjsip: New configuration setting to
allow disabling norefersub
(Reported by Dan Cropp)
* ASTERISK-28320 - Added ARI resource
/ari/channels/{channelid}/rtp_statistics
(Reported by
sungtae kim)
* ASTERISK-28267 - res_stasis: Add ability to switch
applications
(Reported by Benjamin Keith Ford)
* ASTERISK-28087 - add flag to allow CALLERID(num) to be placed
in Contact header in chan_pjsip
(Reported by Torrey
Searle)
* ASTERISK-27971 - res_pjsip: Implement additional SIP RFCs for
Google Voice trunk compatability
(Reported by Nick French)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-30024 - Failed to sign STIR/SHAKEN payload with
functionality not enabled
(Reported by Claude Diderich)
* ASTERISK-29859 - VoiceMailMain() fails when encountering
non-numeric CALLERID(num)
(Reported by Mark Murawski)
* ASTERISK-29816 - SAY_DTMF_INTERRUPT channel variable is not
honored
(Reported by Sean Bright)
* ASTERISK-29821 - Deadlock in bridge_channel_internal_join()
on local channels.
(Reported by Krzysztof Trempala)
* ASTERISK-29779 - progdocs: Hidden code sections with syntax
errors.
(Reported by Alexander Traud)
* ASTERISK-29732 - progdocs: Fix grouping for latest Doxygen
(Reported by Alexander Traud)
* ASTERISK-29771 - Crash occurs when 2 realtime sippeers mysql
connections are configured and we have a schema warning
(Reported by Mario Ban)
* ASTERISK-29776 - stir/shaken: Requires GNU designator
(Reported by Alexander Traud)
* ASTERISK-29764 - chan_misdn: Fix for Doxygen
(Reported
by Alexander Traud)
* ASTERISK-29773 - progdocs: doxyref.h outdated
(Reported
by Alexander Traud)
* ASTERISK-29765 - xmldoc: Fix for Doxygen
(Reported by
Alexander Traud)
* ASTERISK-29730 - Segfault in __ao2_ref if refdebug = yes
(Reported by Alexei Gradinari)
* ASTERISK-29762 - channels: Fix for Doxygen
(Reported by
Alexander Traud)
* ASTERISK-29748 - bridging: Infinite loop when both Local
channel halves in same bridge
(Reported by Joshua C. Colp)
* ASTERISK-29754 - odbc: Fix for Doxygen
(Reported by
Alexander Traud)
* ASTERISK-29753 - parking: Fix for Doxygen
(Reported by
Alexander Traud)
* ASTERISK-29755 - frame: Fix for Doxygen
(Reported by
Alexander Traud)
* ASTERISK-29756 - res_ari: Fix for Doxygen
(Reported by
Alexander Traud)
* ASTERISK-29751 - channel: Fix for Doxygen
(Reported by
Alexander Traud)
* ASTERISK-29750 - stasis: Fix for Doxygen
(Reported by
Alexander Traud)
* ASTERISK-29752 - app: Fix for Doxygen
(Reported by
Alexander Traud)
* ASTERISK-29749 - res_xmpp: Fix for Doxygen
(Reported by
Alexander Traud)
* ASTERISK-29742 - addons: Fix for Doxygen.
(Reported by
Alexander Traud)
* ASTERISK-29747 - res_pjsip: Fix for Doxygen
(Reported
by Alexander Traud)
* ASTERISK-29737 - chan_iax2: Fix for Doxygen
(Reported
by Alexander Traud)
* ASTERISK-29743 - bridges: Fix for Doxygen
(Reported by
Alexander Traud)
* ASTERISK-29741 - tests: Fix for Doxygen
(Reported by
Alexander Traud)
* ASTERISK-29740 - apps: Fix for Doxygen
(Reported by
Alexander Traud)
* ASTERISK-29733 - progdocs: Avoid name with Doxygen \file
(Reported by Alexander Traud)
* ASTERISK-29736 - bridge_channel: Fix for Doxygen
(Reported by Alexander Traud)
* ASTERISK-29735 - progdocs: Avoid multiple use of section
labels
(Reported by Alexander Traud)
* ASTERISK-29734 - progdocs: Use Doxygen \example correctly
(Reported by Alexander Traud)
* ASTERISK-29744 - app_morsecode: Fix deadlock
(Reported
by N A)
* ASTERISK-29703 - res_pjsip_callerid: Fix OLI parsing
(Reported by N A)
* ASTERISK-29705 - app_read: Fix custom terminator
functionality regression
(Reported by N A)
* ASTERISK-29724 - BuildSystem: In POSIX sh, == in place of is undefined.
(Reported by Alexander Traud)
* ASTERISK-29702 - sig_analog: Fix truncated buffer copy
(Reported by N A)
* ASTERISK-28040 - pbx: "dialplan reload" is removing minus
symbol from dynamic hints
(Reported by Daniel Zanutti)
* ASTERISK-29391 - VoiceMail does not cancel recording on
rerecord hangup
(Reported by N A)
* ASTERISK-29709 - res_snmp: Not build on recent Debian
distributions.
(Reported by Alexander Traud)
* ASTERISK-29710 - stasis: Clang 13 warns about the unused but
set variable dispatched.
(Reported by Alexander Traud)
* ASTERISK-29711 - aelparse: GCC 11.2 found two maybe
uninitialized
(Reported by Alexander Traud)
* ASTERISK-29713 - GCC 11.2: two stringop-overread
(Reported by Alexander Traud)
* ASTERISK-29682 - Squash compiler issues generated by gcc 11
(Reported by George Joseph)
* ASTERISK-29693 - Using --with-crypto and --with-ssl fails on
a recompile
(Reported by George Joseph)
* ASTERISK-27816 - func_talkdetect's logic is completely
broken
(Reported by Moritz Fain)
* ASTERISK-29691 - stun: Not all users provide a dst to
ast_stun_request
(Reported by Dennis Haney)
* ASTERISK-26497 - make install downloads x86_32 variants of
external modules on non Intel architectures
(Reported by
Corey Farrell)
* 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-29635 - MP3Player don' t work with actual mpg123
versions
(Reported by Carlos Oliva)
* ASTERISK-29654 - pjproject includes trailing whitespace in
sdp format attributes
(Reported by George Joseph)
* 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-29507 - STUN timeout is silently delaying calls
(Reported by S��bastien Duthil)
* ASTERISK-27871 - Remote URL in playback must end with file
extension
(Reported by Caesar)
* 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-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-29372 - file.c switch does not account for flash
events
(Reported by N A)
* ASTERISK-29358 - chan_pjsip: Trace message for progress is
output even if frame is not queued
(Reported by Michael
Maier)
* ASTERISK-29407 - chan_local: Filtering audio formats should
not occur on removed streams
(Reported by Joshua C. Colp)
* ASTERISK-29030 - res_rtp_asterisk: Additional RTP-frame (with
wrong SSRC) gets inserted when switching from progress to
established
(Reported by Matthias Hensler)
* 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-24631 - Incorrect description of option "context" in
queues.conf.sample
(Reported by Etienne Lessard)
* ASTERISK-26614 - app_queue: updatecdr option in queues.conf
does effectively nothing
(Reported by Alexander Gonchiy)
* 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-29293 - res_config_pgsql: Limit realtime_pgsql() to
return one (no more) record
(Reported by Boris P. Korzun)
* ASTERISK-29303 - pjsip: Re-invite occurs when it shouldn't
(Reported by Benjamin Keith Ford)
* ASTERISK-29311 - res_odbc_transaction sets forcecommit
default value based on isolation level instead of forcecommit
(Reported by Jaco Kroon)
* 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-29287 - app.h: C++ compatibility broken
(Reported by Jean Aunis - Prescom)
* ASTERISK-28369 - app_queue: Member device state "invalid"
when second call is ringing and hint is used
(Reported by
Boolah )
* 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-29261 - res_pjsip: user=phone validation fail for
isup numbers containing *#
(Reported by Mark Petersen)
* ASTERISK-29259 - channel: Allow text+video media streams,
again.
(Reported by Alexander Traud)
* ASTERISK-29258 - chan_sip: Audio stream rejected, Other
stream present: Invalid SDP.
(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-29248 - res_pjsip_session: res sometimes
uninitialized reported by compiler Clang.
(Reported by
Alexander Traud)
* ASTERISK-29229 - Stasis/messaging: text messages not
dispatched to all subscribers when using generic subscription
(Reported by Jean Aunis - Prescom)
* ASTERISK-29240 - chan_pjsip: Incoming PJSIP calls set global
SIPDOMAIN instead of a channel variable
(Reported by Ivan
Poddubny)
* 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-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-29201 - Crash occurs when Transfer and execute
Hangup before the Transfer result
(Reported by Dan Cropp)
* ASTERISK-28947 - Segmentation fault in mixmonitor_ds_destroy
(Reported by Robert Sutton)
* 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-29024 - pjsip: Route Header in Cancel request
incorrectly set
(Reported by Flole Systems)
* ASTERISK-29209 - Debug messages printed by scope trace might
be missing newlines
(Reported by Alexander Traud)
* 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-29145 - GCC Warnings with OPTIMIZE=-Os make
(Reported by Alexander Traud)
* ASTERISK-29146 - GCC Warnings: ���%s��� directive argument is
null.
(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-26424 - app_voicemail: Undocumented behavior from
VMSayName
(Reported by Eric Smith)
* ASTERISK-29097 - res_pjsip_config_wizard: Crash when freeing
string when failing to add extension
(Reported by Vieri)
* ASTERISK-29091 - Crash when ast_translator_build_path fails
(Reported by Jasper van der Neut)
* 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-28311 - dsp: ast_dsp_silence_noise_with_energy wrong
judgment of frame format
(Reported by ���������)
* ASTERISK-24329 - Music On Hold announcement cuts intro of
music the first time it is played
(Reported by Thomas
Frederiksen)
* ASTERISK-29085 - func_curl: Segmentation fault when using
CURL after setting httpheader CURLOPT
(Reported by P��ter
Juh��sz)
* 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-29034 - Lastpause of realtime members is reseting
(Reported by Evandro C��sar Arruda)
* 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-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)
* ASTERISK-28975 - res_http_websocket: Text payload data
doesn't necessary include trailing zero
(Reported by
Nickolay V. Shmyrev)
* ASTERISK-28951 - Inconsistent behaviour queues.conf when
there is (not) a [general] section
(Reported by Walter
Doekes)
* ASTERISK-28965 - res_pjsip: Apply outbound proxy to static
contacts on AOR
(Reported by Joshua C. Colp)
* ASTERISK-28930 - ./configure --without-ssl build failure
(Reported by Jaco Kroon)
* ASTERISK-28957 - chan_sip: chan_sip does not process 400
response to an INVITE.
(Reported by Frederic LE FOLL)
* ASTERISK-28886 - chan_pjsip: PJSIP_SC_NULL does not exist in
pjproject 2.7.2
(Reported by Jared Smith)
* ASTERISK-28888 - res_corosync: causes asterisk crash in huge
distributed environment.
(Reported by Universit�� di
Bologna - CESIA VoIP)
* ASTERISK-28954 - StreamEcho() only returns 1 active stream
(Reported by Bill Kervaski)
* ASTERISK-28955 - "setvar" doesn't work properly in
dahdi-channels.conf
(Reported by Marin Odrljin)
* ASTERISK-28953 - res_pjsip_session: Preserve stream label
(Reported by Joshua C. Colp)
* ASTERISK-28942 - res_sorcery_memory_cache: Individual object
expiration behaves unexpectedly with full backend caching
(Reported by Joshua C. Colp)
* ASTERISK-28950 - Stale code in app_queue to check untouched
channel
(Reported by Walter Doekes)
* ASTERISK-28644 - Stale comment in app_queue about ring_entry
exception
(Reported by Walter Doekes)
* ASTERISK-28952 - Queue wrapuptime sometimes not respected
(based on stale lastcall time)
(Reported by Walter Doekes)
* ASTERISK-28938 - core_unreal / core_local: Add support for
multistream and re-negotiation
(Reported by Joshua C.
Colp)
* ASTERISK-28948 - ARI channel create doesn't referencing the
channel_id parameter
(Reported by sungtae kim)
* ASTERISK-28939 - res_rtp_asterisk: Don't have send/receive
buffers on non-WebRTC
(Reported by Joshua C. Colp)
* ASTERISK-28944 - bridge_softmix: Transitioning a stream from
inactive -> sendrecv/sendonly doesn't re-negotiation
(Reported by Joshua C. Colp)
* ASTERISK-28923 - T.38 Segfaults in chan_pjsip_queryoption
(Reported by Yury Kirsanov)
* ASTERISK-28940 - /channels/create doesn't get any parameters
from the body
(Reported by sungtae kim)
* ASTERISK-28936 - res_pjsip: crash when dialing non-sip uri
(Reported by Walter Doekes)
* ASTERISK-28900 - res_fax: Double frame free when gateway in
use with off-nominal format usage
(Reported by Gregory
Massel)
* ASTERISK-28929 - pjproject_bundled: Honor
--without-pjproject.
(Reported by Alexander Traud)
* ASTERISK-28932 - res_pjsip_logger writing too big packets
(Reported by nappsoft)
* ASTERISK-28920 - bridge show all causes crash
(Reported
by sungtae kim)
* ASTERISK-28921 - Wrong return value check for fwrite when
writing to pcap file
(Reported by nappsoft)
* ASTERISK-28794 - res_pjsip: Crash when escaping during URI
printing
(Reported by nappsoft)
* ASTERISK-28884 - x-ast-orig-host not filtered out from
request URI and To header
(Reported by nappsoft)
* ASTERISK-28871 - res_pjsip_session: Unnecessary re-Invite on
call answer
(Reported by Alexei Gradinari)
* ASTERISK-28903 - res_srtp: Answered Crypto Suite might be
wrong in SDP/SDES.
(Reported by Alexander Traud)
* ASTERISK-28898 - bridge_softmix: Conference bridge not
passing silent rtp packets
(Reported by Jonathan Hunter)
* ASTERISK-28892 - res_musiconhold: Module res_musiconhold
throws false warning
(Reported by Nicholas John Koch)
* ASTERISK-28904 - RTP ICE leaks the memory
(Reported by
sungtae kim)
* ASTERISK-26780 - res_pjsip: PJSIP Registration Fails when
transport=transport-udp6
(Reported by Peter Sokolov)
* ASTERISK-28854 - SIGSEGV when pjsip show history encounters
IPV6 address
(Reported by Roger James)
* ASTERISK-28797 - [patch] tcptls: Fix notice when TLS is
enabled but not configured.
(Reported by Alexander Traud)
* ASTERISK-28804 - [patch] app_osplookup.c: Avoid a format
truncation.
(Reported by Alexander Traud)
* ASTERISK-28776 - Non async-signal-safe syscalls used after
fork before exec
(Reported by nappsoft)
* ASTERISK-28870 - streams: One memory leak and one issue
cloning streams
(Reported by George Joseph)
* ASTERISK-28829 - app_queue: leaking stasis subscription when
Redirecting call
(Reported by laszlovl)
* ASTERISK-25844 - app_queue: Ghost channels in "core show
channels" output
(Reported by Etienne Lessard)
* ASTERISK-28859 - pjsip: Increase maximum candidate count
(Reported by Joshua C. Colp)
* ASTERISK-22920 - Crash while Forwarding from TLS extension
with CHANNEL args secure_bridge_media and
secure_bridge_signaling
(Reported by Shlomi Gutman)
* ASTERISK-28852 - Unprotected access to nochecksums variable,
causes build failures
(Reported by Guido Falsi)
* ASTERISK-28848 - app_fax: Compile.
(Reported by
Alexander Traud)
* ASTERISK-28846 - stream: Enforce formats immutability
(Reported by Joshua C. Colp)
* ASTERISK-28847 - ARI channels cuts the endpoint string over
80 characters
(Reported by sungtae kim)
* ASTERISK-28811 - Crash occurs when fax session switches from
T.38 to audio
(Reported by Alexey Vasilyev)
* ASTERISK-28839 - Sporadic crashes with Segmentation fault
(Reported by Joeran Vinzens)
* ASTERISK-28835 - IPv6 addresses in SDP incorrectly formatted
(Reported by Daniel Heckl)
* ASTERISK-28372 - Asterisk REPLY Wrong Contact header port
(TCP)
(Reported by Anton Satskiy)
* ASTERISK-24428 - Document that Asterisk will use the default
SIP ports (5060 for TCP, 5061 for TLS) if the extern option
variants aren't used
(Reported by sstream)
* ASTERISK-28838 - AST_MODULE_INFO requires, MODULEINFO does
not mention
(Reported by Alexander Traud)
* ASTERISK-28841 - app_confbridge: Add support for disabling
text messaging for a user
(Reported by Joshua C. Colp)
* ASTERISK-28837 - pjproject_bundled: Honor
--without-pjproject.
(Reported by Alexander Traud)
* ASTERISK-28827 - res_rtp_asterisk: Loop when receive buffer
is flushed by a received packet that is also in receive buffer
with NACK
(Reported by nappsoft)
* ASTERISK-27195 - chan_sip: only sets ToS bits on UDP socket,
ignoring TCP and TLS sockets
(Reported by Joshua Roys)
* ASTERISK-28826 - res_rtp_asterisk: Duplicate seqnos being
added to send buffer with NACK
(Reported by nappsoft)
* ASTERISK-28812 - First DTMF is not get
(Reported by
Bernard Merindol)
* ASTERISK-28758 - pjsip startup errors when using "with-ssl"
configure option
(Reported by Patrick Wakano)
* ASTERISK-28824 - BuildSystem: Search for Python/C API when
possibly needed only.
(Reported by Alexander Traud)
* ASTERISK-27717 - [patch] BuildSystem: In NetBSD, the Python
Programming Language is python-2.7.
(Reported by Alexander
Traud)
* ASTERISK-28817 - chan_pjsip: constant DTMF tone if RTP is not
setup yet
(Reported by Kevin Harwell)
* ASTERISK-28819 - [patch] bridge_softmix_binaural: Show state
in menuselect.
(Reported by Alexander Traud)
* ASTERISK-28816 - [patch] BuildSystem: Remove doc/tex and
doc/pdf leftovers.
(Reported by Alexander Traud)
* ASTERISK-28818 - [patch] BuildSystem: Allow space in path.
(Reported by Alexander Traud)
* ASTERISK-28809 - [patch] res_rtp_asterisk: Avoid absolute
value on unsigned subtraction.
(Reported by Alexander
Traud)
* ASTERISK-28796 - func_channel: cannot read fields exten,
context, userfield, channame from dialplan
(Reported by
S��bastien Duthil)
* ASTERISK-28803 - [patch] chan_unistim: Avoid tautological
warnings with clang.
(Reported by Alexander Traud)
* ASTERISK-28808 - [patch] test_stasis: Avoid always true
warning with clang.
(Reported by Alexander Traud)
* ASTERISK-28056 - res_pjsip: Incorrect endpoint status after
endpoint synchronization for a specific AOR
(Reported by
Jason Hord)
* ASTERISK-28795 - channel: write to a stream on multi-frame
writes
(Reported by Kevin Harwell)
* ASTERISK-28789 - test_utils: incorrectly printing error
'declined to load'
(Reported by Alexander Traud)
* ASTERISK-28788 - func_aes: incorrectly printing error
'declined to load'
(Reported by Alexander Traud)
* ASTERISK-28790 - Crash during conference call using
confbridge and video
(Reported by Pascal Cadotte Michaud)
* ASTERISK-16676 - DAHDIRAS fails to properly initiate pppd
unless asterisk is running as root
(Reported by Jaco
Kroon)
* ASTERISK-21205 - [patch] dundi_read_result crash due to
negative number
(Reported by Jaco Kroon)
* ASTERISK-28784 - res_pjsip_sdp_rtp: Only do hold/unhold on
first audio stream
(Reported by Joshua C. Colp)
* ASTERISK-28743 - Asterisk is crashing if the 200 OK with SDP
(Reported by sungtae kim)
* ASTERISK-28783 - res_pjsip_session: Allow default non-audio
streams to have reflected state
(Reported by Joshua C.
Colp)
* ASTERISK-28774 - chan_pjsip's rtptimeout is erroneously
triggered during direct-media (native_rtp) bridge
(Reported by Michael Neuhauser)
* ASTERISK-20325 - Comments in configs/func_odbc.conf.sample
are not consistent with examples. Missing examples.
(Reported by Olivier Krief)
* ASTERISK-28780 - app_mixmonitor: Memory leak due to race
condition between AMI MixMonitor and hangup
(Reported by
Joshua C. Colp)
* ASTERISK-28773 - Incorrect Sender SSRC in RTCP when p2p rtp
bridge is active
(Reported by Torrey Searle)
* ASTERISK-28769 - DTLS Handshake Fails to Occur if ice_support
is enabled but not used
(Reported by Torrey Searle)
* ASTERISK-28759 - A non negotiated rtp frame causes call
disconnection when there is a SSRC change
(Reported by
Paulo Vicentini)
* ASTERISK-26711 - func_enum: ENUM code wrong case
(Reported by Vitold)
* ASTERISK-23407 - Fix the FSF address in the headers of lots
of pjproject files
(Reported by Jared Smith)
* ASTERISK-19460 - [patch] Function TXTCIDNAME never actually
makes DNS calls and always returns an empty string
(Reported by George Joseph)
* ASTERISK-28766 - PJSIP blind transfer not completed after
using Proceeding()
(Reported by laszlovl)
* ASTERISK-28764 - res_rtp_asterisk: Improve NACK support and
seqno handling
(Reported by Joshua C. Colp)
* ASTERISK-28755 - SIP/Stasis: SIP headers not transmitted in
the "variables" field
(Reported by Jean Aunis - Prescom)
* ASTERISK-28685 - check_expr2: linking (when hardening) and
cross-compiling troubles
(Reported by Sebastian Kemper)
* ASTERISK-28754 - ASTERISK-28738 Causes Audio Issue After
Hold
(Reported by Ross Beer)
* ASTERISK-28697 - res_pjsip: Named ACL does not update on
reload if changed
(Reported by Timothy Vanderaerden)
* ASTERISK-28746 - res_pjsip_outbound_registration keeps
retrying the first entry in a SRV record set
(Reported by
George Joseph)
* ASTERISK-28716 - ICE: pjnath shouldn't wait for ICE to
complete before allowing sending
(Reported by Benjamin
Keith Ford)
* ASTERISK-28738 - Incorrect state machine used when
MOH_PASSTHRU is used
(Reported by Torrey Searle)
* ASTERISK-28742 - res_rtp_asterisk: static for audio due to
incomplete dtls/srtp setup
(Reported by Kevin Harwell)
* ASTERISK-28735 - Realtime MoH Unknown format '' -- defaulting
to SLIN
(Reported by Ross Beer)
* ASTERISK-28730 - res_pjsip_session: Fix out of order session
refreshes
(Reported by Joshua C. Colp)
* ASTERISK-26955 - pjsip: SIP Packets with Via "received="
Containing IPv6 Address Delimited by "[]" Rejected
(Reported by Peter Sokolov)
* ASTERISK-28718 - chan_sip: Returns 403 if RTP ports are
depleted, should return 503
(Reported by Walter Doekes)
* ASTERISK-28713 - res_stasis_playback: Error building JSON
(Reported by S��bastien Duthil)
* ASTERISK-28714 - REGRESSION: Feature
subscription_persistence_recreate (ASTERISK-27759) Causes
Segfaults
(Reported by Ross Beer)
* ASTERISK-26082 - res_pjsip_messaging: MessageSend
Content-Type can't be changed
(Reported by Alex)
* ASTERISK-28423 - ARI causes STASIS Deadlock
(Reported
by Ross Beer)
* ASTERISK-28679 - stasis application is destroyed after its
creation
(Reported by Francois Blackburn)
* ASTERISK-25421 - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in
spite of the error when sending
(Reported by Dmitriy
Serov)
* ASTERISK-28686 - chan_sip strictrtp=yes fails when media
source is changed: no audio
(Reported by Walter Doekes)
* ASTERISK-28139 - RTP Stream Incorrect Payload Type Causes
Asterisk To Drop Calls
(Reported by Paul Brooks)
* ASTERISK-28677 - CDR billsec is always 0 for transferred
calls
(Reported by Maciej Michno)
* ASTERISK-28702 - chan_dahdi: holding a channel via flash to
dialtone times out after 0:16:40
(Reported by Andrew
Siplas)
* ASTERISK-24484 - Update documentation for statsd module -
usage requirements unclear
(Reported by Dan Jenkins)
* ASTERISK-28706 - silk 24hHz doesn't show up in 'core show
translation' output
(Reported by Sean Bright)
* ASTERISK-28695 - core: minmemfree watermark uses free RAM,
not available RAM
(Reported by Kevin Flyn)
* ASTERISK-28693 - chan_sip: SIP MESSAGE beginning with a
whitespace appears empty in the dialplan
(Reported by
Frank Matano)
* ASTERISK-23739 - [patch]Segfault forwarding voicemail with
ODBC storage enabled and realtime voicemail_data is used
(Reported by Stas Kobzar)
* ASTERISK-27622 - empty voicemail.conf required for ARA
(realtime) voicemail to leave message
(Reported by Jim Van
Meggelen)
* ASTERISK-21794 - CLI command 'realtime update2' syntax
failure when using according to usage help
(Reported by
Cedric BASSAGET)
* ASTERISK-28349 - Pause reason not reported in QueueMember AMI
event
(Reported by Niksa Baldun)
* ASTERISK-25429 - res_pjsip_endpoint_identifier_ip: Document
support for hostnames
(Reported by Joshua C. Colp)
* ASTERISK-27775 - res_pjsip_notify: Multiple Event headers can
be present instead of just one
(Reported by
AvayaXAsterisk)
* ASTERISK-28682 - app_record: Lack of `beep` audio file causes
application to return error and hangup
(Reported by Corey
Farrell)
* ASTERISK-28507 - Wiki docs missing for MessageWaiting
(Reported by David M. Lee)
* ASTERISK-27759 - res_pjsip_pubsub: Subscription persistence
does not preserve XML <dialog-info> version number
(Reported by Bryan Nelson)
* ASTERISK-28605 - chan_dahdi: Deadlock in Hangup Scenarios
with concurrent command pri show span X
(Reported by Dirk
Wendland)
* ASTERISK-28633 - stasis bridge topic leak
(Reported by
Joeran Vinzens)
* ASTERISK-28492 - pjsip reload not reloading wizard
endpoint/pickup_group endpoint/call_group
(Reported by
Jean-Denis Girard)
* ASTERISK-28562 - SIP WSS message not processed until next
frame arrives
(Reported by Robert Sutton)
* ASTERISK-28667 - Asterisk ignores parsing of config files if
a Byte order mark is present
(Reported by Robin Leffmann)
* ASTERISK-28625 - Playback of local files impacted by large
media cache
(Reported by Kevin Reeves)
* ASTERISK-27243 - contrib: valgrind.supp doesn't suppress what
it's supposed to due to invalid syntax
(Reported by
Richard Kenner)
* ASTERISK-28664 - "trustrpid" is misspelled in
sip_to_pjsip.py
(Reported by Pascal Cadotte Michaud)
* ASTERISK-28636 - app_chanisavail+cdr: ChanIsAvail sometimes
fails to deactivate CDR.
(Reported by Frederic LE FOLL)
* ASTERISK-28604 - app_meetme, chan_ooh323 and cdr_mysql don't
build on 17.0.0
(Reported by George Joseph)
* ASTERISK-28659 - res_pjsip_sdp_rtp: Bundle includes
non-existent media stream if codecs create additional streams
and offer does not have them
(Reported by nappsoft)
* ASTERISK-28660 - res_fax: wrap Asterisk initiated negotiation
with config option
(Reported by Kevin Harwell)
* ASTERISK-28626 - Missing arguments in PJSIP_CONTACT function
documentation
(Reported by Pascal Cadotte Michaud)
* ASTERISK-28609 - Memory Leak in res_rtp_asterisk.c
(Reported by Ted G)
* ASTERISK-28651 - chan_sip logs errors on tx to non-existent
TCP connections
(Reported by Jaco Kroon)
* ASTERISK-28502 - chan_pjsip incorrectly re-writes REGISTER
200 Response Contact
(Reported by Ross Beer)
* ASTERISK-28641 - res_pjsip Segfaults when realtime
configuration to an AOR points to a not existent AOR
(Reported by Ross Beer)
* ASTERISK-28647 - chan_sip: RTP frames not transmitted after
emitting a COLP
(Reported by Jean Aunis - Prescom)
* ASTERISK-28637 - chan_sip+native_bridge_rtp: directmedia
compatibility check failure when negociated ptime is not default
ptime.
(Reported by Frederic LE FOLL)
* ASTERISK-28445 - res_pjsip_session: ast_json_vpack: Invalid
UTF-8 string on hangup when TEST_FRAMEWORK enabled
(Reported by Bernhard Schmidt)
* ASTERISK-28631 - res_parking: Doesn't park when parkee and
parker are the same
(Reported by Ross Beer)
* ASTERISK-28621 - Enforce T.38 error correction mode at 200 ok
received
(Reported by Salah Ahmed)
* ASTERISK-28624 - res_pjsip_outbound_registration: add SRV
failover
(Reported by Kevin Harwell)
* ASTERISK-28608 - app_amd: Use time calculation to calculate
timeout
(Reported by Michael Cargile)
* ASTERISK-28615 - chan_dahdi: PRI span status may stay "Down,
Active" after a short alarm
(Reported by Frederic LE FOLL)
* ASTERISK-28576 - res_rtp_asterisk: ICE Completion Crash when
sent packet length doesn't match
(Reported by Joshua
Elson)
* ASTERISK-26481 - FILE function grabs garbage along with read
data when target line has no newline
(Reported by Jonathan
Harris)
* ASTERISK-28618 - bridge_softmix: hold not cleared when
joining a softmix bridge
(Reported by Kevin Harwell)
* ASTERISK-28616 - parking: Deadlock when multi call parking
(Reported by Joshua C. Colp)
* ASTERISK-28572 - Memory leaks in res_calendar_exchange and
res_calendar_icalendar
(Reported by Yoooooo Ha)
* ASTERISK-28585 - ari/resource_events: Crash in event session
cleanup
(Reported by Kevin Harwell)
* ASTERISK-28590 - utils.c throws repeated warnings;
"pthread_attr_setstacksize: Invalid argument"
(Reported by
Speed Dial Dave)
* ASTERISK-28578 - race condition on pjsip channelstats
command
(Reported by Salah Ahmed)
* ASTERISK-28571 - cdr_pgsql: accesses obsolete (and finally
removed) column
(Reported by Christoph Moench-Tegeder)
* ASTERISK-28575 - MWI Send Notify Crash on 16.6
(Reported by Joshua Elson)
* ASTERISK-28574 - pjproject fails to build on 16.6.0, works on
16.5
(Reported by Niklas Larsson)
* ASTERISK-28561 - Asterisk Deadlocks
(Reported by
Aheliotech)
* ASTERISK-28086 - chan_pjsip: Crash when initiating PlayDTMF
over AMI
(Reported by Jeremiah Gadd)
* ASTERISK-28552 - res_pjsip_mwi: Frack during unload on
unsolicited_mwi container
(Reported by Kevin Harwell)
* ASTERISK-28566 - CDR backend unload problem during active
call(s)
(Reported by Marian Piater)
* ASTERISK-28553 - stasis.c: Crash during unload
(Reported by Kevin Harwell)
* ASTERISK-28544 - Wrong contact representation in ipv6 mode
(Reported by J��rgen H)
* ASTERISK-28534 - Segmentation fault when there is no priority
for an extension
(Reported by Timothy Vanderaerden)
* ASTERISK-28463 - res_pjsip_path: Crash when invalid contact
is configured
(Reported by Juan Martin)
* ASTERISK-28521 - pjsip: Memory Leak
(Reported by Mark)
* ASTERISK-28523 - Asterisk 16.5.0 Memory leak
(Reported
by Cyril Rami��re)
* ASTERISK-28536 - Asterisk release candidates fail to build on
FreeBSD
(Reported by Guido Falsi)
* ASTERISK-28538 - chan_pjsip: Deadlock on fax detection
(Reported by Joshua C. Colp)
* ASTERISK-28497 - func_odbc: truncating Unicode string on
readsql
(Reported by Boris P. Korzun)
* ASTERISK-23756 - setvar directive when used in template and a
child of said template, results in duplicate variable names
(Reported by Michael Goryainov)
* ASTERISK-28527 - ChanIsAvail() creates a CDR if
unanswered=yes is set in cdr.conf
(Reported by Frederic LE
FOLL)
* ASTERISK-28525 - chan_dahdi: set CHANNEL(hangupsource) when a
PRI channel hangs up
(Reported by Frederic LE FOLL)
* ASTERISK-28511 - codec_resample: Bad sound quality when up
sampling from SLIN16 to SLIN32
(Reported by Ruddy G)
* ASTERISK-28499 - translate: Crash when frame does not have a
"src" field set
(Reported by Gregory Massel)
* ASTERISK-25592 - chan_unistim: Clang Warning: variable sized
type not at end of a struct
(Reported by Alexander Traud)
* ASTERISK-28488 - pjsip mwi: n+1 sip notify's sent on
re-register
(Reported by Chris Savinovich)
* ASTERISK-28509 - PJSIP cnonce generated on Linux contains 36
characters, NEC only supports up to 32 characters
(Reported by Dan Cropp)
* ASTERISK-28505 - app_voicemail/IMAP: segfault in
leave_voicemail because not checking mailstream
(Reported
by Alexei Gradinari)
* ASTERISK-28487 - compile menuselect on gentoo
(Reported
by Kilburn)
* ASTERISK-28472 - Asterisk occasionally passes a NULL as
srtp->session to srtp_protect/unprotect causing SEGV
(Reported by Jonas Swiatek)
* ASTERISK-28498 - cel / cdr: Event times may be incorrect
(Reported by Joshua C. Colp)
* ASTERISK-28480 - json integer overflow in ssrc and timestamp
(Reported by Salah Ahmed)
* ASTERISK-28228 - res_pjsip: pjsip show contacts prints double
entries
(Reported by Ian Jones)
* ASTERISK-28483 - packet lost on UDPTL wrap around
(Reported by Torrey Searle)
* ASTERISK-28477 - Crash when not specifying "dbfile" in
res_config_sqlite3.conf
(Reported by Dennis)
* ASTERISK-28478 - Crash performing "core reload" with modified
res_config_sqlite3.conf
(Reported by Dennis)
* ASTERISK-28282 - AST_SCHED_REPLACE_UNREF causes wait-on-self
deadlocks (in chan_sip)
(Reported by Walter Doekes)
* ASTERISK-27121 - res_pjsip_mwi: Memory leak on reload
(Reported by Sergej Kasumovic)
* ASTERISK-28457 - [patch] Fix crash in chan_dahdi on 32-bit
systems caused by ASTERISK-28317
(Reported by abelbeck)
* ASTERISK-28458 - res_pjsip_sdp_rtp: Remove unused variable
(Reported by Michael Maier)
* ASTERISK-26006 - Show offending IP for TLS setup failures in
logs
(Reported by Oleksandr Natalenko)
* ASTERISK-28444 - chan_pjsip: Peer IP for SSL handshake errors
not logged
(Reported by Bernhard Schmidt)
* ASTERISK-26968 - chan_pjsip: Transfer() does not result in
TRANSFERSTATUS reflecting SIP response to transfer
(Reported by Dan Cropp)
* ASTERISK-28419 - app_amd: Does not work with silence
suppression
(Reported by Nasir Iqbal)
* ASTERISK-28018 - IP Fragmentation happening instead of DTLS
fragmentation on handshake server hello certificate
(Reported by vijay kumar)
* ASTERISK-25371 - Crash in hangup at chan_pjsip.c:1749 when
Asterisk attempts to generate hangup event
(Reported by
Abhay Gupta)
* ASTERISK-28435 - cdr_pgsql: Unix socket doesn't work
(Reported by Dmitry Svyatogorov)
* ASTERISK-27981 - res_fax: Fax session leak with fax
gatewaying
(Reported by pasandev)
* ASTERISK-28427 - new mwi.h include missing from some dahdi
source files, causes build failure
(Reported by Guido
Falsi)
* ASTERISK-28421 - Wrong type used for timestamp in
res_rtp_asterisk
(Reported by Morten Tryfoss)
* ASTERISK-28161 - Removal of Previous Patch Causes PJSIP Timer
Issues
(Reported by Ross Beer)
* ASTERISK-27994 - PJSIP: Early media ringback not indicated
after Progress()
(Reported by Gregory Massel)
* ASTERISK-28412 - GCC 9 catches more string formatting issues
(Reported by George Joseph)
* ASTERISK-28379 - pjsip: show channelstats incorrect
information output
(Reported by Vyrva Igor)
* ASTERISK-28399 - channel.c: Exceptionally long queue length
queuing
(Reported by Abhay Gupta)
* ASTERISK-28392 - The no-partial-inlining flag isn't passed to
the bundled pjproject or jansson builds
(Reported by
George Joseph)
* ASTERISK-28402 - res_pjsip_registrar: SEGV in
registrar_find_contact
(Reported by Ross Beer)
* ASTERISK-27756 - bridge: Failure to impart a channel results
in bad data causing crash
(Reported by Abhay Gupta)
* ASTERISK-26718 - ARI: Bridge destroying doesn't work as
expected
(Reported by Marin Odrljin)
* ASTERISK-28143 - app_amd: Infinite loop on silent calls
(Reported by Abhay Gupta)
* ASTERISK-28353 - stasis: Crash at shutdown when statistics
enabled
(Reported by Joshua C. Colp)
* ASTERISK-28374 - latest asterisk unconditionally launch gcc
--version, even if the compiler is different
(Reported by
Guido Falsi)
* ASTERISK-28391 - res_indications: Crash requesting
autocomplete on indications cli command
(Reported by Lucas
Mendes)
* ASTERISK-27935 - app_voicemail: emailbody per user can't
contain commas
(Reported by S��bastien Duthil)
* ASTERISK-17695 - 1.8.3.2 extenpatternmatchnew=yes cannot find
extensions with '-' in them
(Reported by test011)
* ASTERISK-17799 - AEL reload causes loss of control in a
macro
(Reported by Kirill Katsnelson)
* ASTERISK-18593 - AEL for loops use Macro app and pipe
delimiter
(Reported by Luke-Jr)
* ASTERISK-14939 - AEL parsers does not find existing label
(Reported by klaus3000)
* ASTERISK-20182 - Parsing a label beginning with a numeric
character in all Goto/GotoIf/GotoIfTime application causes
unexpected behavior
(Reported by Janu)
* ASTERISK-28348 - Failed to initialize OOH323 endpoint-OOH323
Disabled
(Reported by Dmitry Shubin)
* ASTERISK-28371 - chan_pjsip: DTMF Mode auto_info fallback
lead to both inband and info
(Reported by Salah Ahmed)
* ASTERISK-28319 - musl: Crash on startup when loading modules
(Reported by Sebastian Kemper)
* ASTERISK-28362 - strtok_r() makes gcc compile warning
(Reported by sungtae kim)
* ASTERISK-28255 - res_rtp_asterisk: REMB RTCP packet sending
may be incorrect
(Reported by Joshua C. Colp)
* ASTERISK-27541 - app_queue: Queue paused reason was (big
number) secs ago when reason is set
(Reported by C��sar
Benjam��n Garc��a Mart��nez)
* ASTERISK-20986 - QUEUE_MEMBER 's description is inaccurate
(Reported by Olivier Krief)
* ASTERISK-28350 - manager: Stasis backed up due to locking
(Reported by Joshua C. Colp)
* ASTERISK-25792 - chan_sip: qualifygap bounds checking
(Reported by Paul Sandys)
* ASTERISK-28341 - res_config_odbc eliminates empty custom (���@���
prefix) variables
(Reported by Alexei Gradinari)
* ASTERISK-28333 - StasisEnd event makes wrong timestamp value
(Reported by sungtae kim)
* ASTERISK-28306 - res_pjsip_mwi: MWI NOTIFY occasionally takes
minutes to be sent
(Reported by Jared Hull)
* ASTERISK-28332 - Variable ALTCONF ignored when service is
used in Debian
(Reported by Cirillo Ferreira)
* ASTERISK-27964 - app_queue: ring_entry accesses nativeformats
without channel lock or reference
(Reported by Francisco
Seratti)
* ASTERISK-28335 - stasis: Make topic and maybe subscription
names unique and more useful
(Reported by Joshua C. Colp)
* ASTERISK-28321 - res_rtp_asterisk: Fixing possible divide by
zero for rtcp stat calculation
(Reported by sungtae kim)
* ASTERISK-28322 - chan_pjsip: Add option to allow ignoring of
183 without SDP
(Reported by Torrey Searle)
* ASTERISK-28328 - MeetMe global non-admin mute is muting
admins that subsequently join
(Reported by Philip Mott)
* ASTERISK-28168 - app_queue: Adding a blank entry into sql
queue_members crashes asterisk.
(Reported by Michael)
* ASTERISK-28323 - pjsip: sip.conf to pjsip.conf conversion
script fails
(Reported by Guido Weckwerth)
* ASTERISK-28272 - The basic-pbx config samples don't produce a
running asterisk
(Reported by George Joseph)
* ASTERISK-28312 - res_pjsip_diversion: Corrupted SIP Diversion
field after handling a 302 redirect
(Reported by Alex
Odrov)
* ASTERISK-24173 - File menuselect/menuselect_gtk.c has no
license header
(Reported by Jeremy Lain��)
* ASTERISK-28166 - app_voicemail: Asterisk unresponsive after
changing voicemail password with ODBC
(Reported by
Michael)
* ASTERISK-28309 - res_pjsip: Wrong Contact and Via fields with
multiple UDP interfaces
(Reported by Nikolay shakin)
* ASTERISK-27992 - PJSIP: Adding `sends_registrations = yes` to
pjsip_wizard.conf causes crash
(Reported by Jonathan
Harris)
* ASTERISK-28213 - res_pjsip: Threads pile up needlessly when
AOR is blocked
(Reported by Ross Beer)
* ASTERISK-28301 - Allow voicemail boxes to be subscribed to
with a presence event package
(Reported by George Joseph)
* ASTERISK-28303 - res_rtp_asterisk: Interaction between
smoother and DTMF can cause out of order timestamps
(Reported by Torrey Searle)
* ASTERISK-28302 - ARI: "Error destroying mutex" when listing
all ARI applications
(Reported by Stefan Repke)
* ASTERISK-28300 - AST_PBX_MAX_STACK is too low for some
applications
(Reported by George Joseph)
* ASTERISK-28106 - Astricon Feedback: Unable to filter ARI
events when GETting causes overload of events
(Reported by
George Joseph)
* ASTERISK-28284 - switching between native_bridge and
simple_bridge can cause one way audio
(Reported by Torrey
Searle)
* ASTERISK-28251 - CI: Fix CI so it reverifies commit message
changes
(Reported by George Joseph)
* ASTERISK-28277 - database: Add some basic logging
(Reported by Joshua C. Colp)
* ASTERISK-28181 - ari: Originating overwrites channel start
time
(Reported by sungtae kim)
* ASTERISK-28173 - Deadlock in chan_sip handling subscribe
request during res_parking reload
(Reported by Giuseppe
Sucameli)
* ASTERISK-28104 - AstriCon Feedback: Automatically create a 1
line dialplan context for stasis apps
(Reported by George
Joseph)
* ASTERISK-28271 - Opensuse Leap 15 --with-jannson-bundled will
not compile
(Reported by David Wilcox)
* ASTERISK-28238 - PJSIP realtime. getcontext not working with
DUNDI
(Reported by Ray)
* ASTERISK-28263 - codec_opus: errors setting max_playback_rate
and bitrate to "sdp"
(Reported by Gianluca Merlo)
* ASTERISK-28257 - res_http_websocket: PING / PONG opcodes
break data reception
(Reported by Jeremy Lain��)
* ASTERISK-28250 - build: Cross-compilation fails for target
arm-linux-gnueabihf
(Reported by Jean Aunis - Prescom)
* ASTERISK-28252 - HangupHandler manager events are never
thrown
(Reported by Gerald Schnabel)
* ASTERISK-28231 - res_http_websocket: Not responding to
Connection Close Frame (opcode 8)
(Reported by Jeremy
Lain��)
* ASTERISK-28249 - res_monitor: Segfault with
Monitor(wav,file,i)
(Reported by Valentin Vidi��)
* ASTERISK-28244 - stasis: Filter messages at publishing to
AMI/ARI
(Reported by Joshua C. Colp)
* ASTERISK-28197 - stasis: ast_endpoint struct holds the
channel_ids of channels past destruction in certain cases
(Reported by Mohit Dhiman)
* ASTERISK-28230 - res_rtp_asterisk: abs-send-time extension
added with Asterisk 15.5.0 breaks GXV3140 video telephony
(Reported by David Kuehling)
* ASTERISK-28232 - core: RAII using clang use-after-scope
issue
(Reported by Diederik de Groot)
* ASTERISK-28162 - [patch] need to reset DTMF last sequence
number and timestamp on RTP renegotiation
(Reported by
Alexei Gradinari)
* ASTERISK-28225 - app_voicemail: Channel variable
VM_MESSAGEFILE not updated correctly if message marked "urgent"
(Reported by boatright)
* ASTERISK-28218 - app_queue: Asterisk crashes when using Queue
with a pre-dial handler (option b)
(Reported by Mark)
* ASTERISK-28212 - stasis: Statistics broke ABI under developer
mode
(Reported by Joshua C. Colp)
* ASTERISK-28222 - Regression: MWI polling no longer works
(Reported by abelbeck)
* ASTERISK-28221 - Bug in ast_coredumper
(Reported by
Andrew Nagy)
* ASTERISK-28215 - app_voicemail: Leaving voicemail sometimes
doesn't trigger NOTIFYs
(Reported by George Joseph)
* ASTERISK-27959 - [patch] Asterisk 15.4.1 h264 fmtp
negotiation problem
(Reported by David Kuehling)
* ASTERISK-28201 - [patch] confbridge: no announce to the
marked users when they join an empty conference
(Reported
by Alexei Gradinari)
* ASTERISK-28117 - stasis: Add statistics for usage when in
developer mode
(Reported by Joshua C. Colp)
* ASTERISK-28186 - stasis: Filter messages at publishing based
on to_* presence
(Reported by Joshua C. Colp)
* ASTERISK-28194 - chan_sip: Leak using contact ACL
(Reported by Giuseppe Sucameli)
* ASTERISK-28157 - Asterisk crashes when the res_pjsip_*
modules unload
(Reported by sungtae kim)
* ASTERISK-28125 - app_queue: Revert broken queue channel
reference patch
(Reported by laszlovl)
* ASTERISK-27095 - chan_pjsip: When connected_line_method is
set to invite, we're not trying UPDATE
(Reported by George
Joseph)
* ASTERISK-28182 - chan_pjsip: When connected_line_method is
set to invite, asterisk is not trying UPDATE
(Reported by
nappsoft)
* ASTERISK-28151 - app_voicemail: MWI fails with
mailboxes=##@device instead of mailboxes=##@default
(Reported by Ronald Raikes)
* ASTERISK-28119 - stasis: Segment channel snapshot to reduce
creation cost
(Reported by Joshua C. Colp)
* ASTERISK-28102 - stasis: Use implementation specific cache
for channel snapshots
(Reported by Joshua C. Colp)
* ASTERISK-28159 - SIGABRT caused by stack corruption in
hashkeys_read when no matching keys present
(Reported by
Michael Walton)
* ASTERISK-28140 - repeated segmentation faults
(Reported by Eyal Hasson)
* ASTERISK-28103 - stasis: Filter messages at publishing to
reduce work done
(Reported by Joshua C. Colp)
* ASTERISK-28169 - ARI /channels/create handler causes core
dump
(Reported by sungtae kim)
* ASTERISK-28129 - Incorrect Behavior for rewrite_contact when
Re-Invite omits routset
(Reported by Torrey Searle)
* ASTERISK-28158 - Some conditions prevent running of el_end,
break the terminal.
(Reported by Corey Farrell)
* ASTERISK-28110 - rtp: Incorrect Packetization
(Reported
by Robert Cripps)
* ASTERISK-28146 - pbx_config: Only the first [globals] section
is processed.
(Reported by Corey Farrell)
* ASTERISK-28150 - Formatting error in documentation
(Reported by Scott Griepentrog)
* ASTERISK-28081 - chan_sip: Asterisk 12+ chan_sip doesn't
report AST_CEL_PICKUP in handle_invite_replaces
(Reported
by Luit van Drongelen)
* ASTERISK-28137 - res_pjsip_notify: improve realtime
performance on CLI completion on the endpoint
(Reported by
Alexei Gradinari)
* ASTERISK-27980 - Caller ID cannot be changed on Attended
Transfer before dialing out
(Reported by Alexei Gradinari)
* ASTERISK-28107 - app_confbridge: Participant info labels
aren't being added to the SDPs
(Reported by George Joseph)
* ASTERISK-28089 - function ast_sendtext() create RTP realtime
packets with a trailing null byte in the payload
(Reported
by Emmanuel BUU)
* ASTERISK-28076 - bridging: Asterisk crashes when receiving an
empty realtime text frame
(Reported by Emmanuel BUU)
* ASTERISK-28084 - app_queue: QueueMemberStatus Event flooding
AMI
(Reported by Andrej)
* ASTERISK-28077 - res_pjsip: improve realtime performance on
CLI 'pjsip show contacts'
(Reported by Alexei Gradinari)
* ASTERISK-27920 - app_queue: Queue member considered inuse
after immediately hanging up during dialing.
(Reported by
Cao Minh Hiep)
* ASTERISK-26094 - stasis: Playing MOH to bridge with ARI does
not work
(Reported by Cameron)
* ASTERISK-28065 - res_odbc: missing SQL error diagnostic
(Reported by Alexei Gradinari)
* ASTERISK-28057 - chan_sip: SipNotify via AMI behaves
differently to CLI
(Reported by Peter Katzmann)
* ASTERISK-28045 - configure script does not enforce
libunbound2 version
(Reported by Samuel Galarneau)
* ASTERISK-28070 - testsuite: Sniffer assumes pjmedia will use
ports below 10000
(Reported by Joshua C. Colp)
* ASTERISK-27854 - rtp: Crash in off-nominal case where RTP
instance can't be set up
(Reported by Lei Fu)
* ASTERISK-28034 - chan_sip unstable with TLS after asterisk
start or reloads
(Reported by David Hajek)
* ASTERISK-28059 - PJSIP: Update bundled PJPROJECT to version
2.8
(Reported by Joshua C. Colp)
* ASTERISK-28047 - chan_pjsip: Declined video stream is added
when no video codecs configured and session refresh with removed
video stream occurs
(Reported by Will)
* ASTERISK-28033 - AMI event "NewExten" is set to the wrong
class
(Reported by laszlovl)
* ASTERISK-28049 - res_pjproject build failure
(Reported
by Jaco Kroon)
* ASTERISK-28029 - [patch] res_musiconhold : music on hold will
not start if previous hold just reached end of file
(Reported by Frederic LE FOLL)
* ASTERISK-28005 - channel.c: ARI ring only once
(Reported by Hajek Michal)
* ASTERISK-28032 - Realtime queuemembers are not updated during
retry phase
(Reported by laszlovl)
* ASTERISK-27988 - alembic: PJSIP
"mwi_subscribe_replaces_unsolicited" field is integer not
boolean
(Reported by Joshua C. Colp)
* ASTERISK-28020 - res_pjsip_transport_websocket: Properly set
'received' for IPv6
(Reported by Sean Bright)
* ASTERISK-28002 - When T.140 realtime text is negociated, a
lot of debug traces are generated
(Reported by Emmanuel
BUU)
* ASTERISK-27881 - PBX calls via chan_sip TCP trunk now get
authentification error
(Reported by Ian Gilmour)
* ASTERISK-28022 - res_pjsip realtime: uri column in
ps_contacts table can be too short
(Reported by Florian
Floimair)
* ASTERISK-27944 - res_pjsip_t38: Crash receiving 1xx responses
other than 100 before 200 for T.38 reINVITE
(Reported by
Joshua Elson)
* ASTERISK-28007 - rtcp-mux is put in SDP answer regardless of
offer
(Reported by Torrey Searle)
* ASTERISK-27398 - No joint capabilities with video and
audio-only streams
(Reported by Benjamin Keith Ford)
* ASTERISK-27973 - app_queue: QUEUESTATUS = CONTINUE instead
LEAVEEMPTY
(Reported by Valentin Safonov)
* ASTERISK-27997 - pjproject_bundled: Fix for Solaris builds.
Do not undef s_addr.
(Reported by Alexander Traud)
* ASTERISK-27999 - Wrong SRTP use status report
(Reported
by Salah Ahmed)
* ASTERISK-28001 - res_pjsip_registrar: Improve performance of
inbound handling
(Reported by Joshua C. Colp)
* ASTERISK-27966 - pjsip: Race condition in 183 re transmission
can result in a deadlock
(Reported by Torrey Searle)
* ASTERISK-15331 - make menuselect fails due to undefined
symbols (initscr32, w32addch) in menuselect_curses.o
(Reported by Majdi Bsoul)
* ASTERISK-14935 - [regression] menuselect compilation failure
on Solaris 10
(Reported by Samuel Owens)
* ASTERISK-12382 - menuselect compilation failure on Solaris 10
/ gcc 3.4.3
(Reported by rleasure)
* ASTERISK-9107 - menuselect compilation failure on Solaris
10/gcc-4.1.1
(Reported by Bob Atkins)
* ASTERISK-27991 - BuildSystem: Enable Jansson in Solaris 11.
(Reported by Alexander Traud)
* ASTERISK-27548 - res_pjsip_endpoint_identifier_ip only
matches against "generic string" headers
(Reported by
George Joseph)
* ASTERISK-27990 - res_rtp_asterisk: Requires OpenSSL in
Developer Mode.
(Reported by Alexander Traud)
* ASTERISK-27591 - Frack errors in stasis.c and memory leakage
(Reported by Siruja Maharjan)
* ASTERISK-27978 - res_pjsip: Change default transport
keepalive to preserve behavior
(Reported by Joshua C.
Colp)
* ASTERISK-27968 - systemd: asterisk.service
(Reported by
seanchann.zhou)
Improvements made in this release:
-----------------------------------
* ASTERISK-29777 - documentation: Standardize example syntax
(Reported by N A)
* ASTERISK-29715 - app_voicemail: Refactor email generation
functions
(Reported by N A)
* ASTERISK-29727 - Add type for JSON stasis message RTCP Report
Received/Sent
(Reported by Boris P. Korzun)
* ASTERISK-29714 - Spelling errors
(Reported by Josh
Soref)
* ASTERISK-29707 - chan_iax2: Allow both key and secret to be
specified at dial time
(Reported by N A)
* ASTERISK-29662 - Add mix option to Playback application for
say and filename
(Reported by Shloime Rosenblum)
* 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-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-29459 - Missing configuration from PJSIP to SIP
conversion script
(Reported by N A)
* ASTERISK-29460 - Recognize application/hook-flash in PJSIP
(Reported by N A)
* ASTERISK-29434 - Asterisk reveals pjproject version in STUN
packets
(Reported by Jeremy Lain��)
* ASTERISK-29349 - Silent voicemail option is not completely
silent
(Reported by N A)
* ASTERISK-29380 - Add Flash AMI event to handle flash events
(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-29336 - documentation: Fix inconsistent support
levels
(Reported by Joshua C. Colp)
* ASTERISK-29335 - xml: Embed module information into core XML
documentation.
(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-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)
* ASTERISK-28959 - res_pjsip: Added option for disable rport
parameter set
(Reported by sungtae kim)
* ASTERISK-28958 - Continue reading string when ping received
by websocket
(Reported by Nickolay V. Shmyrev)
* ASTERISK-28945 - AMI SendText - add Content-Type parameter
(Reported by Kevin Harwell)
* ASTERISK-28949 - res_http_websocket: Add masking to websocket
client
(Reported by Moises Silva)
* ASTERISK-28899 - Upgrade Asterisk to bundled pjproject 2.10
(Reported by Kevin Harwell)
* ASTERISK-28895 - res_pjsip_logger: Add tons'o'functionality
(Reported by Joshua C. Colp)
* ASTERISK-28896 - ari: Add support for specifying variables on
channel create
(Reported by Joshua C. Colp)
* ASTERISK-28879 - pjproject has race conditions in it's build
system
(Reported by Guido Falsi)
* ASTERISK-28866 - third-party/pjproject/configure.m4 contains
bashisms
(Reported by Guido Falsi)
* ASTERISK-28853 - Missing include on FreeBSD
(Reported
by Guido Falsi)
* ASTERISK-28832 - chan_mobile creates PCMA streams that make
some VoIP clients crash or not render received audio
(Reported by Peter Turczak)
* ASTERISK-28813 - func_volume: Allow decimal numbers as
parameter to improve granularity
(Reported by Jean Aunis -
Prescom)
* ASTERISK-28777 - Codec Negotiation: add
outgoing_call_offer_prefs option
(Reported by Kevin
Harwell)
* ASTERISK-27946 - dial (API): Storage of dialed target uses
AST_MAX_EXTENSION when it shouldn't
(Reported by Joshua
Elson)
* ASTERISK-28782 - Add support for Content-Disposition header
in multi-part INVITES
(Reported by Torrey Searle)
* ASTERISK-28787 - res_pjsip_session: Decide more intelligently
when to add video
(Reported by Joshua C. Colp)
* ASTERISK-28756 - Codec Negotiation: add
incoming_call_offer_pref option
(Reported by Kevin
Harwell)
* ASTERISK-28750 - TLS/SSL Key too small error
(Reported
by Martin Zeh)
* ASTERISK-28733 - stream: Add support for adding/removing
streams during SFU/calls
(Reported by Joshua C. Colp)
* ASTERISK-24798 - Documentation - Clarify That Format Is Set
By File Name Extension In MixMonitor
(Reported by xrobau)
* ASTERISK-28726 - install_prereq script uses the interactive
mode when installing aptitude
(Reported by Sylvain
Afchain)
* ASTERISK-28710 - Should be able to disable the /httpstatus
URI in the built-in HTTP server
(Reported by Sean Bright)
* ASTERISK-28484 - Add AudioSocket support
(Reported by
Se��n C. McCord)
* ASTERISK-28638 - Simplify dialplan for Dial, Page, and
ChanIsAvail
(Reported by cmaj)
* ASTERISK-28673 - GET FULL VARIABLE documentation
clarification
(Reported by Jonathan Harris)
* ASTERISK-28629 - [patch] Add an "inhibitCOLP" flag to the
bridges REST API
(Reported by Jean Aunis - Prescom)
* ASTERISK-28658 - app_confbridge: Add support for setting
maximum sample rate
(Reported by Joshua C. Colp)
* ASTERISK-28602 - res_pjsip_outbound_registration: Maximum
retries reached
(Reported by Daniel)
* ASTERISK-28586 - Typo in README-SERIOUSLY.bestpractices.md
(Reported by Sam Banks)
* ASTERISK-22192 - [patch] Allow voicemail forwards with ODBC
backend when format differs from attachfmt column
(Reported by cmaj)
* ASTERISK-28567 - Problem with ASTERISK-20207: Asterisk should
clear out any .lock files in the voice mail directory on
startup.
(Reported by Michael)
* ASTERISK-28542 - [patch] add the ability for asterisk to
generate on-hold re-invites
(Reported by Torrey Searle)
* ASTERISK-28512 - Add pass-through support for H.265 (HEVC)
codec
(Reported by Florian Floimair)
* ASTERISK-28443 - app_voicemail: remove dependency on stasis
cache
(Reported by Kevin Harwell)
* ASTERISK-28442 - stasis_state: Create a stasis module to
cache last known state
(Reported by Kevin Harwell)
* ASTERISK-28385 - res_ari_channels: Added detail hangup code
settings
(Reported by sungtae kim)
* ASTERISK-28234 - pbx_dundi: Add IPv4/IPv6 dual bind support
for DUNDi
(Reported by Kirsty Tyerman)
* ASTERISK-28401 - app_confbridge: Add *_all remb behavior
variants
(Reported by Joshua C. Colp)
* ASTERISK-28400 - res_rtp_asterisk / res_pjsip_sdp_rtp: Add
support for transport-cc
(Reported by Joshua C. Colp)
* ASTERISK-28363 - Millisecond-resolution call stats including
PDD in channel variables
(Reported by Antoni Goldstein)
* ASTERISK-28378 - Added detail subscriber/subscription info
for stasis show app cli
(Reported by sungtae kim)
* ASTERISK-20207 - Asterisk should clear out any .lock files in
the voice mail directory on startup.
(Reported by Steven
Wheeler)
* ASTERISK-28111 - build: CHANGES/UPGRADE are irritating to
work with.
(Reported by Corey Farrell)
* ASTERISK-28264 - Added topic_all container
(Reported by
sungtae kim)
* ASTERISK-28343 - Added app_name, app_data to channel type
(Reported by sungtae kim)
* ASTERISK-28326 - ari: Added timestamp for some ari events.
(Reported by sungtae kim)
* ASTERISK-28317 - Add logical group at DAHDIChannel event and
create "dahdi_group" at CHANNEL function
(Reported by
Cirillo Ferreira)
* ASTERISK-28279 - Added creation timestamp for bridge
(Reported by sungtae kim)
* ASTERISK-27483 - Allow wrapuptime to be set for each queue
member
(Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-28055 - app_queue: Per-member wrapup time missing
from AddQueueMember application
(Reported by Niksa Baldun)
* ASTERISK-28292 - Changed to show all channel stats including
wrong media
(Reported by sungtae kim)
* ASTERISK-28253 - res_pjsip_session: Adding rtcp stats result
into the session
(Reported by sungtae kim)
* ASTERISK-28246 - Support skipping on the g726 format
(Reported by Eyal Hasson)
* ASTERISK-28196 - bridge_softmix: Does not support WebRTC
source with multi video tracks.
(Reported by Xiemin Chen)
* ASTERISK-28198 - res_ari: Add new hangup causes for ARI
Channel DELETE command
(Reported by Sebastian Damm)
* ASTERISK-28144 - [patch] New function PJSIP_PARSE_URI to
parse an URI and return a specified part of the URI
(Reported by Alexei Gradinari)
* ASTERISK-28136 - Allow the sip_to_pjsip script to be used in
a pipe
(Reported by Pascal Cadotte Michaud)
* ASTERISK-28046 - Remove stale nonoptreq references
(Reported by Walter Doekes)
* ASTERISK-27164 - [patch] Add IPv6 Support for DUNDi
(Reported by Adam Secombe)
* ASTERISK-28006 - PJSIP: Missing
"party=calling"/"party=called" in Remote-Party-ID
(Reported by Eric Dantie)
* ASTERISK-27995 - pjproject_bundled: Find shared libraries in
root --with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27993 - pjsip_wizard example gives wrong info about
unsupported SRV records
(Reported by Jonathan Harris)
* ASTERISK-27970 - res_rtp_asterisk: T.140 packets containing
backspace or end of line are merged with regular text and it
causes some UA to break
(Reported by Emmanuel BUU)
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-18.9-cert1
Thank you for your continued support of Asterisk!
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