[asterisk-dev] Asterisk 18.0.0-rc1 Now Available
Asterisk Development Team
asteriskteam at digium.com
Wed Sep 9 12:14:13 CDT 2020
The Asterisk Development Team would like to announce the first
release candidate of Asterisk 18.0.0.
This release candidate is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 18.0.0-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:
Security bugs fixed in this release:
-----------------------------------
* 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)
New Features made in this release:
-----------------------------------
* 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 lvl)
* 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)
Bugs fixed in this release:
-----------------------------------
* 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 lvl)
* 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 lvl)
* 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)
Improvements made in this release:
-----------------------------------
* 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)
For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0-rc1
Thank you for your continued support of Asterisk!
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