[asterisk-dev] Asterisk 17.0.0-rc1 Now Available
Asterisk Development Team
asteriskteam at digium.com
Wed Aug 28 13:10:38 CDT 2019
The Asterisk Development Team would like to announce the first
release candidate of Asterisk 17.0.0.
This release candidate is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 17.0.0-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-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-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-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-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 lvl)
* 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 lvl)
* 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 lvl)
* 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-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 candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0-rc1
Thank you for your continued support of Asterisk!
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