Asterisk Development Team
2019-Oct-28 11:18 UTC
[asterisk-users] Asterisk 17.0.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk
17.0.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 17.0.0 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-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-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-28561 - Asterisk Deadlocks
(Reported by
Aheliotech)
* 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-28521 - pjsip: Memory Leak
(Reported by Mark)
* ASTERISK-28523 - Asterisk 16.5.0 Memory leak
(Reported
by Cyril Rami��re)
* ASTERISK-28538 - chan_pjsip: Deadlock on fax detection
(Reported by Joshua C. Colp)
* ASTERISK-28536 - Asterisk release candidates fail to build on
FreeBSD
(Reported by Guido Falsi)
* ASTERISK-28499 - translate: Crash when frame does not have a
"src" field set
(Reported by Gregory Massel)
* 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-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, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
Thank you for your continued support of Asterisk!
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