Asterisk Development Team
2020-Oct-20 12:00 UTC
[asterisk-users] Asterisk 18.0.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk
18.0.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 18.0.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
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-29109 - res_pjsip_session: Asterisk 18 does not
progress calls due to codec negotiation after upgrading from
Asterisk 16
(Reported by Ross Beer)
* 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, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
Thank you for your continued support of Asterisk!
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20201020/52c600ff/attachment-0001.html>
Hello! On 20.10.20 at 14:00 Asterisk Development Team wrote:> The Asterisk Development Team would like to announce the release of Asterisk 18.0.0. > This release is available for immediate download at > https://downloads.asterisk.org/pub/telephony/asteriskI just tested the new codec negotiation feature and unfortunately wasn't able to get it working as expected. I tried several configurations - but none has been working - the result has always been the same. Use case: Alice calls Bob - sends INVITE G722 / alaw / ulaw Configured in Asterisk for this device: G722 / alaw / ulaw / gsm A: codec_prefs_incoming_offer = prefer: configured, operation: intersect, keep: all, transcode: prevent Bob: Configured in Asterisk for this device: alaw / ulaw B: codec_prefs_outgoing_offer = prefer: configured or pending, operation: intersect, keep: first or all, transcode: prevent Asterisk sends INVITE to Bob alaw / ulaw Asterisk receives OK from Bob alaw B: codec_prefs_incoming_answer = prefer: configured or pending, operation: intersect, keep: first or all, transcode: prevent Asterisk sends OK to Alice G722 / alaw / ulaw A: codec_prefs_outgoing_answer = prefer: pending, operation: intersect, keep: first or all, transcode: prevent => I would have expected alaw to be sent to A - but G722 / alaw / ulaw is sent and transcoding is active! What did I do wrong? Could you please add the correct configuration you expect to get the expected result alaw? Thanks Kind regards Michael
On Wed, Oct 21, 2020 at 7:46 AM Michael Maier <m1278468 at mailbox.org> wrote:> Hello! > > On 20.10.20 at 14:00 Asterisk Development Team wrote: > > The Asterisk Development Team would like to announce the release of > Asterisk 18.0.0. > > This release is available for immediate download at > > https://downloads.asterisk.org/pub/telephony/asterisk > > I just tested the new codec negotiation feature and unfortunately wasn't > able to get it working as expected. I tried several configurations - but > none has been working - the result > has always been the same. >This is expected right now. Foundational aspects were put in, but there is still work to be done for PJSIP which will land in a future release. The complexity of it and the investigation of how things work, interactions, etc took considerably longer than expected. If there's specific scenarios that you'd like to ensure are met you can reach out on the asterisk-dev mailing list and George Joseph will add them to the list if not already present. Even just from AstriDevCon there were some things that individuals brought up. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20201021/8c68d1e4/attachment.html>
Possibly Parallel Threads
- Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
- set codec based on B side
- Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
- Asterisk 18.0.0 Now Available
- Asterisk 18.0.0 Now Available