Asterisk Development Team
2018-08-08 21:17:32 UTC
The Asterisk Development Team would like to announce the first
release candidate of Asterisk 16.0.0.
This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.0.0-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-27807 - iostreams: Potential DoS when client
connection closed prematurely
(Reported by Sean Bright)
* ASTERISK-27818 - Username bruteforce is possible when using
ACL with PJSIP
(Reported by John)
* ASTERISK-27658 - WebSocket frames with 0 sized payload causes
DoS
(Reported by Sean Bright)
* ASTERISK-27583 - Segmentation fault occurs in asterisk with
an invalid SDP fmtp attribute
(Reported by Sandro Gauci)
* ASTERISK-27582 - Segmentation fault occurs in Asterisk with
an invalid SDP media format description
(Reported by
Sandro Gauci)
* ASTERISK-27618 - Crash occurs when sending a repeated number
of INVITE messages over TCP or TLS transport
(Reported by
Sandro Gauci)
* ASTERISK-27640 - SUBSCRIBE message with a large Accept value
causes stack corruption
(Reported by Sandro Gauci)
New Features made in this release:
-----------------------------------
* ASTERISK-27286 - Add the ability to read the media file type
from HTTP header for playback
(Reported by Gaurav Khurana)
* ASTERISK-27704 - Add cache_pools debug option to
pjproject.conf
(Reported by Richard Mudgett)
* ASTERISK-27581 - Add new AMI Action for PJSIPShowContacts
(Reported by sungtae kim)
* ASTERISK-27547 - res_pjsip: Add new AMI Action for
PJSIPShowAuths
(Reported by sungtae kim)
* ASTERISK-27117 - core: Add support for timelen parsing to
ast_parse_arg and ACO.
(Reported by Corey Farrell)
* ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
incoming INVITE Request-URI.
(Reported by Richard Mudgett)
* ASTERISK-27413 - Add cache_media_frames debugging option.
(Reported by Richard Mudgett)
* ASTERISK-27206 - res_pjsip: No mechanism exists to limit
endpoint identification to IP only
(Reported by Ben
Merrills)
* ASTERISK-27215 - [patch]AMI : Add CancelAtxfer Action
(Reported by Thomas Sevestre)
* ASTERISK-27322 - [New Feature] Add mute and DTMF passthrough
to ARI add channel to bridge
(Reported by Darren Sessions)
* ASTERISK-27162 - [patch]chan_sip: Access incoming SIP REFER
headers in the dialplan
(Reported by Kirill Katsnelson)
* ASTERISK-27163 - chan_sip: Dialplan function SIP_HEADERS() to
complement SIP_HEADER().
(Reported by Kirill Katsnelson)
* ASTERISK-27063 - Add support for systemd socket activation
(Reported by Corey Farrell)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27978 - res_pjsip: Change default transport
keepalive to preserve behavior
(Reported by Joshua Colp)
* ASTERISK-27880 - [patch] pjproject_bundled: Repair
./configure --with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27810 - BASIC-RETRANS: Implement receive
(Reported by Benjamin Keith Ford)
* ASTERISK-27972 - res_sorcery_config: Allow object name based
matching
(Reported by Joshua Colp)
* ASTERISK-27965 - module: Remove old modules, update support
levels
(Reported by Joshua Colp)
* ASTERISK-25548 - stasis: Improve message type "Use of before
init/after destruction" error
(Reported by Joshua Colp)
* ASTERISK-27967 - srtp: rejecting short sdes lifetimes
incompatible with obihai ATAs
(Reported by Nick French)
* ASTERISK-27961 - res_pjsip: Spurious ERROR logging when
printing headers in sip_msg
(Reported by Nick French)
* ASTERISK-27563 - pjsip modules always get -O2 even when
DONT_OPTIMIZE is set
(Reported by George Joseph)
* ASTERISK-27347 - [patch] pjproject_bundled: Disable TCP/TLS
keep-alives.
(Reported by Alexander Traud)
* ASTERISK-27957 - PJSIP proposes ICE candidates on answer even
if not in offer
(Reported by Torrey Searle)
* ASTERISK-27938 - [patch] Compile fails with `IPTOS_MINCOST'
undeclared.
(Reported by Alexander Traud)
* ASTERISK-27955 - res_pjsip_session: sdp group:BUNDLE
attribute truncated
(Reported by Kevin Harwell)
* ASTERISK-27956 - res_pjsip_pubsub: segfault in function
publish_expire
(Reported by Alexei Gradinari)
* ASTERISK-27949 - res_pjsip_rfc3326: A lot of endpoints do not
correctly handle two Reason headers
(Reported by Ross
Beer)
* ASTERISK-27763 - res_pjsip_session: Initial INVITE with
audio+fax results in 488 instead of declining stream
(Reported by Thiago Coutinho)
* ASTERISK-27657 - res_pjsip_t38: ATA fails with hangupcause
58(Bearer capability not available)
(Reported by Jared
Hull)
* ASTERISK-27080 - res_pjsip_t38: Slow T.38 re-invite rejection
if remote leg has T.38 disabled
(Reported by Torrey
Searle)
* ASTERISK-26686 - res_pjsip: Lock inversion in transport
management
(Reported by Ross Beer)
* ASTERISK-27939 - [patch] bridge_softmix_binaural: Enable
FFTW3 in Solaris 11.
(Reported by Alexander Traud)
* ASTERISK-27783 - res_pjsip_pubsub: apparent crash on
shutdown
(Reported by Kevin Harwell)
* ASTERISK-27870 - app_confbridge: Conference bridge and
announcer channels are not removed if conference is ended as
soon as it starts
(Reported by Robert Mordec)
* ASTERISK-27909 - cdr: Deadlock with submit_scheduled_batch
and submit_unscheduled_batch
(Reported by Denis Lebedev)
* ASTERISK-26987 - pbx_dundi: Asterisk crashes when unloading
module pbx_dundi.so with dundi peers
(Reported by Kirsty
Tyerman)
* ASTERISK-27943 - AMI: Action SendText needs to use the
correct thread.
(Reported by Richard Mudgett)
* ASTERISK-27942 - res_pjsip_messaging doesn't accept
application/* content-types.
(Reported by George Joseph)
* ASTERISK-27936 - res_pjsip_session doesn't update media when
a 200 comes in with a different port than a 183
(Reported
by George Joseph)
* ASTERISK-27933 - [patch] uuid: Enable UUID in Solaris 11.
(Reported by Alexander Traud)
* ASTERISK-27625 - channels: CHECK_BLOCKING is ineffective
(Reported by Corey Farrell)
* ASTERISK-27931 - [patch] BuildSystem: Enable ./configure in
Solaris 11.
(Reported by Alexander Traud)
* ASTERISK-27926 - [patch] bootstrap.sh: find -maxdepth is not
POSIX compatible.
(Reported by Alexander Traud)
* ASTERISK-27903 - menuselect: GCC 8: restrict-qualified
parameter passed and aliased.
(Reported by Alexander
Traud)
* ASTERISK-27914 - [patch] tests/test_utils: Repair ./configure
--with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27705 - chan_iax2: Stops listening for traffic
(Reported by Kirsty Tyerman)
* ASTERISK-27848 - rtp: DTMF Breaks With telephony-event/16000
(Reported by Dominic)
* ASTERISK-27908 - [patch] crypto.h: Repair ./configure
--with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27905 - [patch] res_srtp: Repair ./configure
--with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27888 - SQL fetch error on query which return 0
columns
(Reported by Alexei Gradinari)
* ASTERISK-27902 - chan_pjsip isn't updating hangupcause on 4XX
responses
(Reported by George Joseph)
* ASTERISK-27901 - [patch] ooh323c: GCC 8: output truncated
before terminating nul.
(Reported by Alexander Traud)
* ASTERISK-27872 - res_pjsip: Modified qualify_frequency
doesn't effect until pjsip reload
(Reported by Alexei
Gradinari)
* ASTERISK-27094 - res_fax: Deadlock when using Local channels
and fax gateway
(Reported by David Brillert)
* ASTERISK-25261 - Manager events for MeetMe have incorrectly
documented key name 'Usernum' - should be 'User'
(Reported
by Francois Blackburn)
* ASTERISK-27878 - [patch] tcptls.h: Repair ./configure
--with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27876 - [patch] tcptls: Allow OpenSSL configured
with no-dh.
(Reported by Alexander Traud)
* ASTERISK-27874 - [patch] tcptls: Allow OpenSSL 1.1.x
configured with enable-ssl3-method no-deprecated.
(Reported by Alexander Traud)
* ASTERISK-27845 - Codec-Change Re-INVITE during DTMF can cause
marker bit error
(Reported by Torrey Searle)
* ASTERISK-27831 - res_rtp_asterisk: Add support for
abs-send-time RTP extension
(Reported by Joshua Colp)
* ASTERISK-27863 - config/ast_destroy_realtime_fields:
successful DELETE is treated as failed
(Reported by Alexei
Gradinari)
* ASTERISK-27865 - [patch]: tcptls: Repair ./configure
--with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27760 - Asterisk ODBC Voicemail Prompt storage fails
with recent MariaDB version.
(Reported by Nic Colledge)
* ASTERISK-27853 - Incorrect error reported when
leaving/retrieving a ODBC voicemail
(Reported by Nic
Colledge)
* ASTERISK-27726 - chan_mobile: presents incorrect inbound
Caller-ID names
(Reported by Brian)
* ASTERISK-27861 - [patch] res_pjsip_endpoint_identifier_ip:
Unregister the module for headers.
(Reported by Alexander
Traud)
* ASTERISK-27852 - cli: "manager show settings" mislabels HTTP
timeout as being minutes.
(Reported by Corey Farrell)
* ASTERISK-27824 - Fix issues exposed by GCC 8
(Reported
by George Joseph)
* ASTERISK-27850 - [patch] rtp_engine: Allow Media Formats with
add_static_payload(-1) on egress again.
(Reported by
Alexander Traud)
* ASTERISK-27811 - [patch] sip_to_pjsip: Enable python3
compatibility.
(Reported by Alexander Traud)
* ASTERISK-27841 - digest over for manager (ami) over http
fails on too long uris
(Reported by Jaco Kroon)
* ASTERISK-26570 - Macro allows an infinite loop of dialplan
inclusion resulting in a crash
(Reported by Tzafrir Cohen)
* ASTERISK-27572 - cdr_mysql creates empty records if
reconnects when mysql was not up on module load
(Reported
by Tzafrir Cohen)
* ASTERISK-27801 - Asterisk got stuck while enabling "ari set
debug all on"
(Reported by shaurya jain)
* ASTERISK-27795 - chan_sip: one way / no audio with srtp
(Reported by Florian Kaiser)
* ASTERISK-27800 - One way audio when calling from Asterisk(sip
trunk) to another number where both are connected to a SBC using
TLS+SRTP
(Reported by Artur Pires)
* ASTERISK-26806 - pjsip_options: rework to make more
efficient
(Reported by Kevin Harwell)
* ASTERISK-27814 - translate: interpolated frames are not
passed through
(Reported by Kevin Harwell)
* ASTERISK-27812 - When the ooh323 debug is on there is no
ringing signal to incoming calls via H323 trunk.
(Reported
by Dimos)
* ASTERISK-26893 - No "alert" or "progress" in chan_ooh323 if
debug is enabled only on the module
(Reported by Marco
Giordani)
* ASTERISK-27804 - bridge_softmix / app_confbridge: Add support
for combining REMB reports
(Reported by Joshua Colp)
* ASTERISK-27639 - [patch] BuildSystem: Enable IMAP storage on
FreeBSD and DragonFly BSD.
(Reported by Alexander Traud)
* ASTERISK-27418 - app_confbridge: "core show profile bridge"
does not output "sfu" when video_mode is sfu
(Reported by
Carlos Chavez)
* ASTERISK-27809 - [patch] utils/pval: Add -lBlocksRuntime for
compiler clang conditionally.
(Reported by Alexander
Traud)
* ASTERISK-27808 - [patch] chan_vpb: Avoid GNU old-style field
designator extension.
(Reported by Alexander Traud)
* ASTERISK-27806 - BASIC-RETRANS: Implement send
(Reported by Benjamin Keith Ford)
* ASTERISK-27774 - res_musiconhold: Music on hold restarts
after every announcement
(Reported by lvl)
* ASTERISK-27782 - cdr_mysql: Missing MYSQL_PORT definition
(Reported by Evandro César Arruda)
* ASTERISK-27614 - res_pjsip_session: SDP origin does not use
resolved address
(Reported by John M.)
* ASTERISK-27776 - res_rtp_asterisk: Add support for sending
RTCP feedback messages
(Reported by Joshua Colp)
* ASTERISK-27740 - chan_sip: New Channel creation from new SIP
dialog with Replaces failed to be properly tracked and
destroyed
(Reported by Shannon Price)
* ASTERISK-27786 - app_confbridge: Add ability to enable and
configure REMB support
(Reported by Joshua Colp)
* ASTERISK-27706 - PJSIP: Deadlock shutting down subscription
TCP connection and sending subscription message.
(Reported
by Ross Beer)
* ASTERISK-27688 - res_pjsip: Crash on TCP PJSIP Transport
Disconnect
(Reported by Ross Beer)
* ASTERISK-27758 - res_rtp_asterisk: Add support for raising
RTCP feedback messages
(Reported by Joshua Colp)
* ASTERISK-26366 - rtp: RTCP messages with REMB trigger fast
picture update
(Reported by Joshua Colp)
* ASTERISK-27773 - Command line not being parsed correctly with
getopt not from glibc
(Reported by Guido Falsi)
* ASTERISK-27435 - [patch] configure:
pjsip_evsub_set_uas_timeout not found.
(Reported by
Alexander Traud)
* ASTERISK-27761 - [patch] BuildSystem: With external editline,
do not require libs for internal editline.
(Reported by
Alexander Traud)
* ASTERISK-27755 - ConfBridge: raise ConfbridgeTalking when put
on hold and clear talking status
(Reported by Kevin
Harwell)
* ASTERISK-27743 - Generic PLC doesn't work if the 2 codecs on
a channel are equal
(Reported by George Joseph)
* ASTERISK-27745 - [patch] BuildSystem: Remove unused
dependency on libltdl.
(Reported by Alexander Traud)
* ASTERISK-12841 - [patch] Make format_ogg_vorbis work on
OpenBSD
(Reported by Michiel van Baak)
* ASTERISK-27720 - [patch] BuildSystem: Enable Advanced Linux
Sound Architecture (ALSA) in NetBSD.
(Reported by
Alexander Traud)
* ASTERISK-27741 - res_pjsip_rfc3326.c
rfc3326_use_reason_header doesn't account for more than one
'Reason' header
(Reported by Ross Beer)
* ASTERISK-27734 - [patch] BuildSystem: Enable IMAP storage on
openSUSE and Arch Linux.
(Reported by Alexander Traud)
* ASTERISK-27686 - [patch] install_prereq: Update FreeBSD
libraries.
(Reported by Alexander Traud)
* ASTERISK-27733 - [patch] res_srtp: Add support for libsrtp2.x
on openSUSE.
(Reported by Alexander Traud)
* ASTERISK-11015 - NetBSD Build Needs RPATH set in 1.2.25
(Reported by Curt Sampson)
* ASTERISK-27641 - BuildSystem: Enable Better Backtraces in
FreeBSD.
(Reported by Alexander Traud)
* ASTERISK-27671 - Deprecate legacy modules
(Reported by
Corey Farrell)
* ASTERISK-25586 - uuid_generate_random detection failure
(Reported by John Nemeth)
* ASTERISK-27721 - [patch] BuildSystem: Enable PortAudio in
NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27715 - [patch] BuildSystem: AC_PATH_PROG sets to
colon character when not found.
(Reported by Alexander
Traud)
* ASTERISK-27554 - res_pjsip_rfc3326: Order of 'Reason' headers
break many endpoints
(Reported by Ross Beer)
* ASTERISK-27703 - AMI Action VoicemailUsersList returns 0
MessageCount
(Reported by Sébastien Duthil)
* ASTERISK-27674 - chan_sip: RTP framing issues on outgoing
calls
(Reported by Jean Aunis - Prescom)
* ASTERISK-27441 - PJSIP: Forked INVITE SDP negotiation gets
one way audio.
(Reported by lvl)
* ASTERISK-27718 - [patch] BuildSystem: Enable Lua in NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27722 - [patch] BuildSystem: Depend not implicitly
but explicitly on external libraries.
(Reported by
Alexander Traud)
* ASTERISK-27719 - [patch] res_http_post: Enable GMime in
NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27716 - [patch] BuildSystem: Enable autotools in
NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27714 - [patch] chan_unistim: NetBSD has an
incompatible struct in_pktinfo.
(Reported by Alexander
Traud)
* ASTERISK-27713 - [patch] BuildSystem: Cast any intptr_t
explicitly to its proposed type.
(Reported by Alexander
Traud)
* ASTERISK-27712 - [patch] BuildSystem: Detect whether
uselocale(.) is available.
(Reported by Alexander Traud)
* ASTERISK-27711 - [patch] BuildSystem: Avoid re-defining of
pthread_* on NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27710 - [patch] BuildSystem: Install init scripts on
openSUSE Tumbleweed.
(Reported by Alexander Traud)
* ASTERISK-27709 - [patch] BuildSystem: Avoid == for comparison
in ./configure.
(Reported by Alexander Traud)
* ASTERISK-27610 - app_amd.so returning TOOLONG before reaching
the timeout
(Reported by Michael Cargile)
* ASTERISK-26688 - Documentation: voicemail.conf.sample shows
512 limit for emailbody field, however this is only true if
compiled with LOW_MEMORY option
(Reported by Fran Vicente)
* ASTERISK-27568 - PJSIP: Crash during SIP attended transfer.
(Reported by Bryan Walters)
* ASTERISK-27659 - Output from rawman truncated if output is
long enough
(Reported by Bojan NemÄiÄ)
* ASTERISK-27692 - bridging: Sometimes cloning the stream
topology causes a crash
(Reported by Richard Mudgett)
* ASTERISK-27488 - core: If frame with unnegotiated format is
read crash will occur
(Reported by Sébastien Duthil)
* ASTERISK-24488 - Wrong remote identity and target in dialog
package XML in NOTIFY
(Reported by Alejandro Padilla)
* ASTERISK-24386 - Asterisk "doc/lang/language-criteria.txt"
needs update or removal.
(Reported by Rusty Newton)
* ASTERISK-27646 - ICE fails with no candidate nominated
(Reported by Thomas Guebels)
* ASTERISK-27689 - [patch] rtp_engine: Load format name / mime
type in uppercase again.
(Reported by Alexander Traud)
* ASTERISK-27679 - res_pjsip: Endpoint destruction does not
free DTLS configuration
(Reported by Mak Dee)
* ASTERISK-27684 - [patch] install_prereq: Update OpenBSD
libraries.
(Reported by Alexander Traud)
* ASTERISK-27680 - [patch] res_calendar: Specialized calendars
depend on symbols of general calendar.
(Reported by
Alexander Traud)
* ASTERISK-27681 - [patch] BuildSystem: Enable IMAP storage on
OpenBSD.
(Reported by Alexander Traud)
* ASTERISK-27677 - [patch] BuildSystem: Enable system provided
libedit on OpenBSD.
(Reported by Alexander Traud)
* ASTERISK-27670 - [patch] BuildSystem: Remove chan_h323
leftovers.
(Reported by Alexander Traud)
* ASTERISK-27595 - [patch] BuildSystem: Invoke ldconfig with
previous paths.
(Reported by Alexander Traud)
* ASTERISK-27631 - [patch] BuildSystem: Do not warn when bash
is not installed.
(Reported by Alexander Traud)
* ASTERISK-27666 - chan_sip: Crash processing CANCEL request
(Reported by Leandro Dardini)
* ASTERISK-27584 - Internal pjproject build doesn't disable
bcg729
(Reported by Stuart Henderson)
* ASTERISK-27669 - [patch] codecs: Add support for WebRTC iLBC
2.0.
(Reported by Alexander Traud)
* ASTERISK-27634 - Determine if the internal editline and
stdtime libraries are still relevant
(Reported by George
Joseph)
* ASTERISK-27642 - [patch] backtrace: Avoid
-Wlogical-not-parentheses.
(Reported by Alexander Traud)
* ASTERISK-27555 - [patch] install_prereq: Update Debian/Ubuntu
libraries.
(Reported by Alexander Traud)
* ASTERISK-27656 - CDR: Leaking channel snapshots allocated by
stasis_channel.c
(Reported by Kristijan Vrban)
* ASTERISK-27426 - chan_console: cannot read and write at the
same time with alsa backend
(Reported by Tzafrir Cohen)
* ASTERISK-27621 - (null) string tailing after AsyncAGIEnd AMI
event
(Reported by sungtae kim)
* ASTERISK-27652 - Null pointer Crash in PJSIP MWI
(Reported by Joshua Elson)
* ASTERISK-27571 - res_pjsip: If SIP response is received
during shutdown a crash may occur
(Reported by Joshua
Colp)
* ASTERISK-27619 - Build System: Require compiler to provide
built-in support for atomic references.
(Reported by Corey
Farrell)
* ASTERISK-27612 - Subscriptions Persist After Expiration and
TCP/TLS Disconnect
(Reported by Ross Beer)
* ASTERISK-27637 - [patch] BuildSystem: Enable autotools in
FreeBSD.
(Reported by Alexander Traud)
* ASTERISK-27635 - [patch] app_voicemail: Avoid always true
warnings with clang.
(Reported by Alexander Traud)
* ASTERISK-27599 - [patch] install_prereq: Update
RHEL/CentOS/Fedora libraries.
(Reported by Alexander
Traud)
* ASTERISK-26563 - core: macOS devmode build fails: variable
'freeswap' set but not used
(Reported by David M. Lee)
* ASTERISK-27630 - [patch] editline: Avoid shifting a negative
signed value.
(Reported by Alexander Traud)
* ASTERISK-16172 - Problems with siren14 codec; problems with
siren7 sound files.
(Reported by Steve Murphy)
* ASTERISK-16951 - [patch] configure.ac in 1.4.37 broken with
autoconf 2.60
(Reported by Stéphan Kochen)
* ASTERISK-27603 - [patch] install_prereq: Download latest
Jansson.
(Reported by Alexander Traud)
* ASTERISK-27620 - New module loader aborts startup if a
required module declines load.
(Reported by snuffy)
* ASTERISK-27607 - [patch] res_config_mysql: Avoid the header
mysql_version.h.
(Reported by Alexander Traud)
* ASTERISK-24598 - When running
./contrib/scripts/install_prereq install-unpackaged pjproject is
installed in wrong place
(Reported by PowerPBX)
* ASTERISK-27602 - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs
a directory.
(Reported by Alexander Traud)
* ASTERISK-27600 - [patch] BuildSystem: Allow make clean all
again.
(Reported by Alexander Traud)
* ASTERISK-27598 - [patch] install_prereq: Support package
manager DNF.
(Reported by Alexander Traud)
* ASTERISK-26596 - Placing call on hold temporarily locks up
set
(Reported by Igor Goncharovsky)
* ASTERISK-27596 - [patch] BuildSystem: Use the detected name
for MD5 everywhere.
(Reported by Alexander Traud)
* ASTERISK-27594 - [patch] BuildSystem: Invoke install not in
GNU but POSIX style.
(Reported by Alexander Traud)
* ASTERISK-27593 - [patch] BuildSystem: In OpenBSD, xmlstarlet
is xml.
(Reported by Alexander Traud)
* ASTERISK-27592 - [patch] BuildSystem: Detect external library
Lua in version 5.3.
(Reported by Alexander Traud)
* ASTERISK-27491 - res_pjsip_endpoint_identifier_ip only
matches against header if match by ip fails
(Reported by
George Joseph)
* ASTERISK-26832 - res_pjsip: Segfault when calling
pjsip_hdr_print_on in sip_msg.c:581
(Reported by Ross
Beer)
* ASTERISK-27589 - [patch] BuildSystem: Avoid $EUID and use id
-u instead.
(Reported by Alexander Traud)
* ASTERISK-27585 - [patch] BuildSystem: Resolve resolv.h not
via Generic but Particular Header-Check.
(Reported by
Alexander Traud)
* ASTERISK-27575 - menuselect : remove obsolete TRACE_FRAMES
compiler flag
(Reported by Jean Aunis - Prescom)
* ASTERISK-27576 - [patch] res_config_pgsql: Avoid typecasting
an int to unsigned char.
(Reported by Alexander Traud)
* ASTERISK-27560 - [patch] clang 5 does not know
-Wno-format-truncation
(Reported by Alexander Traud)
* ASTERISK-27578 - [patch] app_osplookup.c: Avoid a format
truncation.
(Reported by Alexander Traud)
* ASTERISK-27577 - [patch] chan_ooh323: Avoid typecasting an
int to unsigned short.
(Reported by Alexander Traud)
* ASTERISK-27534 - chan_sip: Assumes iostream is non-NULL when
it may not be
(Reported by Lubos Dolezel)
* ASTERISK-27549 - [patch] translate: Avoid absolute value on
unsigned substraction.
(Reported by Alexander Traud)
* ASTERISK-27566 - res_pjsip_session: Improve WebRTC interop
with bundling during renegotiation
(Reported by Joshua
Colp)
* ASTERISK-27553 - [patch] res_curl: Avoid error message on
unload.
(Reported by Alexander Traud)
* ASTERISK-27557 - [patch] clang 5.0: implicit conversion to
char changes value to negative.
(Reported by Alexander
Traud)
* ASTERISK-27550 - [patch] bridge_softmix: Avoid warning about
an uninitialized variable.
(Reported by Alexander Traud)
* ASTERISK-27559 - [patch] editline: Avoid comparison between
pointer and zero character constant.
(Reported by
Alexander Traud)
* ASTERISK-27558 - [patch] codec_gsm: Avoid shifting a negative
signed value.
(Reported by Alexander Traud)
* ASTERISK-25329 - Asterisk configure fails on 'cannot find
ptlib-config', despite ptlib-config existing
(Reported by
Rusty Newton)
* ASTERISK-27552 - [patch] chan_ooh323: Limit outgoinglimit to
positive values as intended.
(Reported by Alexander Traud)
* ASTERISK-27551 - [patch] ooh323cDriver: Fix typo in header
guard.
(Reported by Alexander Traud)
* ASTERISK-26046 - [patch] Avoid obsolete warnings on
autoconf.
(Reported by Alexander Traud)
* ASTERISK-20346 - Modules need to ensure that any functions,
apps, AMI actions, etc. they register are unregistered if the
module declines loading
(Reported by Mark Michelson)
* ASTERISK-27539 - 'cdr submit' fails: batch mode not enabled.
(Reported by Tzafrir Cohen)
* ASTERISK-27498 - ICE candidate parser - ICE foundation
parsing too short
(Reported by Michele Prà )
* ASTERISK-25128 - Datastore: Implement automatic module
references.
(Reported by Corey Farrell)
* ASTERISK-27366 - Asterisk Turkish Language Set Problem
(Reported by Halil Ä°brahim YILDIZ)
* ASTERISK-23133 - Documentation fix - MASTER_CHANNEL
Unexpected Behaviour
(Reported by Shane Mitchell)
* ASTERISK-27531 - Compiler optimizations can break module load
sequence.
(Reported by abelbeck)
* ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
Contact crashes asterisk
(Reported by Ross Beer)
* ASTERISK-24198 - Typo's
(Reported by Walter Doekes)
* ASTERISK-27229 - bridge: Old channel video source not set to
NULL after unref
(Reported by Richard Kenner)
* ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
(Reported by Corey Farrell)
* ASTERISK-25079 - AMI bridge of channels results in MOH not
destroyed and robotic audio on one channel
(Reported by
Zane Conkle)
* ASTERISK-27490 - chan_console: 'set active' fails to work
(Reported by Tzafrir Cohen)
* ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
read()
(Reported by Abhay Gupta)
* ASTERISK-24756 - ConfBridge sound_muted does not work from
CLI or AMI
(Reported by Thomas Frederiksen)
* ASTERISK-25649 - Transfer application does not work with
Local channels - documentation misleading
(Reported by
Ivan Ullmann)
* ASTERISK-25869 - chan_sip: "rejected because extension not
found" should be logged as a security event
(Reported by
Brian J. Murrell)
* ASTERISK-27440 - Strictrtp has issues to qualify video rtp
streams
(Reported by Wim De Vlaminck)
* ASTERISK-19657 - Coverity Report: Fix issues for error type
CHAR_IO
(Reported by Matt Jordan)
* ASTERISK-27175 - iax.conf demo peer is invalid
(Reported by Tzafrir Cohen)
* ASTERISK-27430 - README refers to security documents that do
not exist.
(Reported by Corey Farrell)
* ASTERISK-20281 - "core set verbose" behaves strangely, can't
alias it, cli.conf example broken
(Reported by Tim
Ringenbach at Asteria Solutions Group)
* ASTERISK-27382 - crash after an invalid rtcp packet from GT48
FXS gateway
(Reported by Tzafrir Cohen)
* ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
RTCP packet will write past where it should
(Reported by
Vitezslav Novy)
* ASTERISK-27408 - Identify causes and fix
pjsip/resolver/srv/failover/in_dialog/transport_tcp
(Reported by Corey Farrell)
* ASTERISK-18411 - Queue members with hints for state_interface
get stuck in "In Use" state.
(Reported by Steven T.
Wheeler)
* ASTERISK-26131 - chan_sip: Crash Asterisk (in
sip_request_call at chan_sip.c) by making a call to a single
character in a dot pattern match
(Reported by Dwayne
Hubbard)
* ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
not applied on reload
(Reported by John Bigelow)
* ASTERISK-27460 - CDR: Deadlock using AMI Originate with
Variable CDR(amaflags)=...
(Reported by Richard Mudgett)
* ASTERISK-27453 - RTP: Blind transfer direct media scenario
results in one way audio.
(Reported by Richard Mudgett)
* ASTERISK-20643 - SIP ICE support - remove hardcoded
limitation on SDP size, make ICE support disabled by default in
SIP, maybe provide a better warning message
(Reported by
Roy)
* ASTERISK-27457 - chan_sip: Guests disallowed via TCP (or TLS)
if existing peer from same IP.
(Reported by Alexander
Traud)
* ASTERISK-26980 - pjsip: Clean up WebRTC disables
(Reported by abelbeck)
* ASTERISK-27452 - Security: chan_skinny: Memory exhaustion if
flooded with unauthenticated requests
(Reported by George
Joseph)
* ASTERISK-27454 - res_http_post: Don't require
GMIME_MAJOR_VERSION
(Reported by Joshua Colp)
* ASTERISK-23735 - Transcoding makes bad choice in high-rate
translations
(Reported by Richard Kenner)
* ASTERISK-27445 - ARI: Updating a bridge gives wrong error
message.
(Reported by Frank Durden)
* ASTERISK-24662 - [patch] column and row headers for Signed
Linear format variants in output of 'core show translation' are
ambiguous
(Reported by Rusty Newton)
* ASTERISK-27353 - H323 audio starts with a delay of 2
seconds.
(Reported by Marco Giordani)
* ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
media
(Reported by Kevin Harwell)
* ASTERISK-27437 - [patch] ICE: server-reflexive candidates
(srflx) with Dual-Stack.
(Reported by Alexander Traud)
* ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
IPv6 addresses.
(Reported by Alexander Traud)
* ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
(Reported by Ivan Larionov)
* ASTERISK-27431 - Asterisk fails to build when openssl headers
are not installed.
(Reported by Corey Farrell)
* ASTERISK-27421 - RTP source learning not working with devices
that have some clock issues
(Reported by nappsoft)
* ASTERISK-27361 - Attended transfer crashes in Asterisk
13.17.2
(Reported by Alessandro Pimenta)
* ASTERISK-27238 - Bridging: Crash freeing a frame that's
already been freed
(Reported by Richard Kenner)
* ASTERISK-27412 - core: Audiohook freeing interpolated frame
when it shouldn't.
(Reported by Mikhail)
* ASTERISK-27423 - app_record: We set the RECORD_STATUS
channel variable before closing the file
(Reported by
George Joseph)
* ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
insert same ip address in "source ip address" and "destination
ip address" fields in HEP packets
(Reported by Max Norba)
* ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
is equal to RemoteAddress)
(Reported by Vasilii Rogin)
* ASTERISK-27415 - asterisk.conf: Setting astctl without
setting astrundir is ineffective.
(Reported by Corey
Farrell)
* ASTERISK-27411 - pjsip: TCP connections may not be destroyed
(Reported by Joshua Colp)
* ASTERISK-27404 - DEBUG_FD_LEAKS does not record socketpair,
timerfd_create or eventfd.
(Reported by Corey Farrell)
* ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
responses.
(Reported by Corey Farrell)
* ASTERISK-27337 - chan_sip: Security vulnerability with client
code header (revisited)
(Reported by Richard Mudgett)
* ASTERISK-27319 - (Security) Function in PJSIP 2.7
miscalculates the length of an unsigned long variable in 64bit
machines
(Reported by Kim youngsung)
* ASTERISK-27391 - Regression: Deadlock between AOR named lock
and pjproject grp lock
(Reported by shaurya jain)
* ASTERISK-27393 - res_pjsip: Crash occurs when an empty
contact read from astdb or database
(Reported by Aaron An)
* ASTERISK-27290 - res_pjsip: PIDF contact field has
malformed/invalid XML
(Reported by basildane)
* ASTERISK-27032 - res_pjsip: TLS options do not handle empty
values
(Reported by seanchann.zhou)
* ASTERISK-27395 - srtp: Add support for ephemeral DTLS
certificates
(Reported by Sean Bright)
* ASTERISK-26426 - format_ogg_opus: remove from source
(Reported by Kevin Harwell)
* ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
enabled but not configured.
(Reported by Alexander Traud)
* ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
(Reported by Alexander Traud)
* ASTERISK-27378 - Modules: Fix issues with CLI completion.
(Reported by Corey Farrell)
* ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
character isn't allowed any more
(Reported by Michael
Maier)
* ASTERISK-27364 - channel: Crash when fax gateway is in use
with PJSIP
(Reported by Jared Hull)
* ASTERISK-27390 - Audit menuselect module dependencies
(Reported by Corey Farrell)
* ASTERISK-27389 - Optional API modules should not allow
unload.
(Reported by Corey Farrell)
* ASTERISK-27369 - Bridge() dialplan application fails without
setting BRIDGERESULT channel variable
(Reported by James
Terhune)
* ASTERISK-27067 - res_ari_channels: channel_state_invalid
always leaks snapshot reference.
(Reported by Marin
Odrljin)
* ASTERISK-27379 - stream: Allow streams on a topology to be
put into groups
(Reported by Joshua Colp)
* ASTERISK-27374 - alembic: PJSIP scripts are missing column
bundle in ps_endpoints table
(Reported by Florian
Floimair)
* ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
documentation
(Reported by Igor Goncharovsky)
* ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
'imap_delete_old_greeting'
(Reported by Anthony Messina)
* ASTERISK-27194 - jitterbuffer: Does not handle case where
translator returns null frame.
(Reported by Joshua Elson)
* ASTERISK-27372 - ARI: Node ARI client broken in latest
versions of 13 and 14
(Reported by Benjamin Keith Ford)
* ASTERISK-26639 - core: Disabling xmldoc support does not
work. Also results in abort during Asterisk startup.
(Reported by Mr Dini)
* ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
absence of the Expires header field with an unsubscribe action.
(Reported by Jonathan Cloots)
* ASTERISK-25960 - The config_hook unit test causes Asterisk to
crash if run a second time
(Reported by George Joseph)
* ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
when rtp_ipv6 set to yes
(Reported by Martin Cisárik)
* ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
curl is loaded
(Reported by Ronald Raikes)
* ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
but first on SDP media level.
(Reported by Alexander
Traud)
* ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
Assertion on un/re-load: mod.id == -1
(Reported by Tzafrir
Cohen)
* ASTERISK-23462 - Cannot disable SIP debugging via CLI after
enabling with conf file option - also 'sip set debug off'
reports debugging disabled, when it really isn't
(Reported
by Rusty Newton)
* ASTERISK-27350 - app_macro deprecation
(Reported by
Corey Farrell)
* ASTERISK-27354 - bridge_softmix: When a channel leaves add in
any missing participant streams
(Reported by Joshua Colp)
* ASTERISK-27333 - sip_to_pjsip not correctly handling
disallow=all directive
(Reported by Torrey Searle)
* ASTERISK-27343 - Fails to build in FreeBSD due to
sys/sysmacros.h not existing there
(Reported by Guido
Falsi)
* ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
(o=) contains local address.
(Reported by Alexander Traud)
* ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
configured codecs, but subset based on caller
(Reported by
lvl)
* ASTERISK-27340 - backtrace.c: Crash due to double-free.
(Reported by Corey Farrell)
* ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
stopping.
(Reported by Alexander Traud)
* ASTERISK-27047 - res_pjsip: user=phone added to Anonymous
caller-id when it shouldn't be.
(Reported by dtryba)
* ASTERISK-26988 - res_pjsip_session: user_eq_phone adds double
user=phone parameters to URIs
(Reported by dtryba)
* ASTERISK-27301 - [patch] app_queue: Music On Hold for
real-time queues is not reset to default
(Reported by
Nathan Bruning)
* ASTERISK-25266 - Application Originate returns SUCCESS to
ORIGINATE_STATUS upon failure to originate
(Reported by
Allen Ford)
* ASTERISK-27270 - cdr_mysql: various crashes at second module
reload if cdr_mysql.conf is configured
(Reported by
Tzafrir Cohen)
* ASTERISK-27328 - Missing openssl dependencies in
res_rtp_asterisk and tcptls
(Reported by Tzafrir Cohen)
* ASTERISK-27192 - res_pjsip: Loss of SIP registrations causing
unavailable endpoints
(Reported by Richard Mudgett)
* ASTERISK-27305 - res_ari: Memory leaks in ARI when using
Content-Type: application/json
(Reported by David Hajek)
* ASTERISK-26922 - chan_sip: tcpbind uses wrong source address
(Reported by Ksenia)
* ASTERISK-27324 - [patch] Dual-Stack server cannot be used as
IPv4 client via TCP/TLS
(Reported by Alexander Traud)
* ASTERISK-27317 - vector: multiple evaluation of elem in
AST_VECTOR_ADD_SORTED.
(Reported by Corey Farrell)
* ASTERISK-27318 - res_pjsip_mwi: uninitialized value from
ast_strings_match
(Reported by Corey Farrell)
* ASTERISK-27284 - Status of RFC 3323 and PJSIP
(Reported
by dtryba)
* ASTERISK-27296 - [patch] False positive busy checks when
icalendar's recurrence-id mechanism is involved
(Reported
by Benoît Dereck-Tricot)
* ASTERISK-27216 - app_queue: does its
check-makeannouncement-logic twice each head-caller-loop
(Reported by Stefan Engström)
* ASTERISK-27298 - Problem with expires on pjsip /
outbound-publish
(Reported by Cyrille Demaret)
* ASTERISK-27295 - Contact is improperly translated after
d178f497
(Reported by Sean Bright)
* ASTERISK-27292 - Multiple RTP Stream Created Breaking RFC2833
(SSRC Changes)
(Reported by Ross Beer)
* ASTERISK-27289 - A codeblock that maintains a bug,but maybe
the codeblock will never run
(Reported by Huangyx)
* ASTERISK-27277 - bridge: Renegotiate if source stream
changes.
(Reported by Joshua Colp)
* ASTERISK-27264 - res_pjsip_session: Crashes after sending
PRACK and receiving 200 OK
(Reported by Daniel Heckl)
* ASTERISK-27283 - Realtime config fail with PostgreSQL version
before 9.1
(Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-27260 - [pjsip] chan_pjsip_indicate: Don't know how
to indicate condition 36
(Reported by Daniel Heckl)
* ASTERISK-27257 - bridge_native_rtp: half-way direct media
when using early bridging
(Reported by Jean Aunis -
Prescom)
* ASTERISK-16898 - SRTP unprotect: authentication failure when
RTP sequence number switches from 65535 -> 0
(Reported by
Marcello Ceschia)
* ASTERISK-27279 - Crash in pubsub_on_rx_request NULL pointer -
Possible PJSIP Vulnerability
(Reported by Ross Beer)
* ASTERISK-25524 - module reload res_calendar.so does not
reload everything in calendar.conf
(Reported by Jesper)
* ASTERISK-27274 - RTCP needs better packet validation to
resist port scans.
(Reported by Richard Mudgett)
* ASTERISK-27252 - RTP: One way audio with direct media and
strictrtp=yes.
(Reported by Richard Mudgett)
* ASTERISK-24588 - res_calendar does not process CalDAV from
Owncloud [fix included]
(Reported by Stefan Gofferje)
* ASTERISK-25523 - res_calendar: Warning about invalid channel
value (for notification) occurs even when event has no
notification configured.
(Reported by Jesper)
* ASTERISK-21399 - RTP Multicast of L16 (type 10): Asterisk and
wireshark disagree
(Reported by Tzafrir Cohen)
* ASTERISK-27248 - [patch]external_media_address and
external_signaling_address don't always honor localnet
(Reported by Walter Doekes)
* ASTERISK-27165 - CDR: CDR(start,u) function won't work in
cdr_custom config
(Reported by Jacek Konieczny)
* ASTERISK-24066 - res_smdi: convert to astobj2
(Reported
by Corey Farrell)
* ASTERISK-27217 - chan_sip: Asterisk crashing when
subscription doesn't get set
(Reported by Bryan Walters)
* ASTERISK-17540 - SDP origin attribute modified when issuing
re-INVITE because of directmedia=yes
(Reported by saghul)
* ASTERISK-27254 - alembic: prune_on_boot fix erroneous
(Reported by Florian Floimair)
* ASTERISK-27232 - When in queue on g722 with interruptions,
music on hold can get stuck and no longer play
(Reported
by Jens T.)
* ASTERISK-27024 - nat/external_media settings ignored in
14.4.1
(Reported by Christopher van de Sande)
* ASTERISK-26879 - PJSIP external_media_address ignored if no
local_net options are provided
(Reported by Matt Jordan)
* ASTERISK-27236 - Segfault ast_channel_name (chan=0x0) at
channel_internal_api.c:478 during T.38 Fax Receive
(Reported by Ross Beer)
* ASTERISK-27225 - Crash when freeing dtls_cfg->cafile
(Reported by Richard Kenner)
* ASTERISK-27177 - ooh323c: misleading indentation in
addons/ooh323c/src/ooSocket.c
(Reported by Tzafrir Cohen)
* ASTERISK-27241 - libc segfault upon entry into app_directory
(Reported by David Moore)
* ASTERISK-27152 - Sending a "tel" uri in a From or To header
in an unauthenticated message causes asterisk to crash
(Reported by Ross Beer)
* ASTERISK-27103 - core: ast_safe_system command injection
possible.
(Reported by Corey Farrell)
* ASTERISK-27013 - res_rtp_asterisk: Media can be hijacked even
with strict RTP enabled
(Reported by Joshua Colp)
* ASTERISK-27231 - res_rtp_asterisk: Allow remote SSRC to
change due to renegotiation
(Reported by Joshua Colp)
* ASTERISK-26994 - Confbridge: CBAnn channels intermittently
become stuck when caller hangs up before recording name
(Reported by James Terhune)
* ASTERISK-27222 - core: Don't queue up multiple video update
frames.
(Reported by Joshua Colp)
* ASTERISK-20858 - app_minivm fails to clean up mkstemp files
(Reported by Walter Doekes)
* ASTERISK-16777 - several filename bugs in Record()
application
(Reported by klaus3000)
* ASTERISK-27168 - alembic: PJSIP scripts are missing column
dtls_fingerprint in ps_endpoints table
(Reported by
Florian Floimair)
* ASTERISK-27209 - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE
is used
(Reported by Torrey Searle)
* ASTERISK-19103 - When using realtime queues, function
QUEUE_MEMBER_LIST() will return an error if no other
app/function has loaded the queues first. This problem does not
exist if queues.conf is used.
(Reported by Jim Van
Meggelen)
* ASTERISK-21241 - When using voicemail as announce only
(maxmsg=0), the star dtmf to enter the voicemail is not honored
(Reported by Eelco Brolman)
* ASTERISK-27212 - bridge_softmix: Quickly joining/leaving may
cause video stream to remain in SFU
(Reported by Richard
Mudgett)
* ASTERISK-27204 - [patch] app_queue: Wrong queue stat
calculation
(Reported by sungtae kim)
* ASTERISK-27207 - XMPP OAuth not working due to inverted
logic
(Reported by Michael Kuron)
* ASTERISK-27174 - res_calendar_icalendar: Recurring events not
being loaded from Google calendar using ical
(Reported by
Mark Thompson)
* ASTERISK-27202 - If wget is not installed and "or" is not
available, external components (excluding pjsip) are not
installed
(Reported by Seán C. McCord)
* ASTERISK-27200 - manager: hook event is not being raised
(Reported by Kevin Harwell)
* ASTERISK-27147 - Either asterisk or pjproject isn't re-using
tcp connections (again)
(Reported by George Joseph)
* ASTERISK-27193 - IPv6 receive address in message doesn't
include brackets
(Reported by Scott Griepentrog)
* ASTERISK-27158 - [patch] res_rtp_asterisk: RTCP statistics
are not available when native bridge is used
(Reported by
Torrey Searle)
* ASTERISK-26745 - Asymmetric codecs when
asymmetric_rtp_codec=no
(Reported by Jesse Ross)
* ASTERISK-27189 - Make --with-pjproject-bundled the default
for Asterisk 15
(Reported by George Joseph)
* ASTERISK-27110 - RTP session is not fully destroyed on
channel hangup
(Reported by Matt Jordan)
* ASTERISK-27182 - bridge: Crash when mapping streams
(Reported by Joshua Colp)
* ASTERISK-27180 - channel: requester leaks joint_cap on
success.
(Reported by Corey Farrell)
* ASTERISK-27179 - res_pjsip_session: Handling of 'msid' is
incorrect
(Reported by Kevin Harwell)
* ASTERISK-27119 - res_pjsip: parse/add msid attribute when
webrtc is enabled
(Reported by Kevin Harwell)
* ASTERISK-27171 - Asterisk 15.0.0-Beta1 does not compile
(Reported by Ira Emus)
* ASTERISK-26659 - res_pjsip: PJSIP presence - missing braces
around the status element in XML
(Reported by Abraham
Liebsch)
* ASTERISK-27156 - Asterisk won't compile on Fedora 26 with
devmode enabled.
(Reported by Corey Farrell)
* ASTERISK-27001 - res_pjsip: TLS connection not stable
(Reported by Ian Gilmour)
* ASTERISK-27130 - Applications ARI: Unsubscribe action for
deviceStates does not remove old subscriptions properly
(Reported by Sergej Kasumovic)
* ASTERISK-25810 - say.c calls for sounds in the subdir
"digits" that don't exist (in Core). SayUnixTime or other Say...
apps will fail out when they call these sounds.
(Reported
by Nicolas Riendeau)
* ASTERISK-27142 - sounds: Conflict between files in
asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
(Reported by Corey Farrell)
* ASTERISK-27143 - bridge_softmix / res_rtp_asterisk: Fix
packet loss and renegotiation issues.
(Reported by Joshua
Colp)
Improvements made in this release:
-----------------------------------
* ASTERISK-22825 - Dialplan Function for Checking Parking Lot
Slot
(Reported by JoshE)
* ASTERISK-27912 - [PATCH] Add predial handler to app_queue
(Reported by Kristian HÞgh)
* ASTERISK-27929 - [patch] BuildSystem: Enable autotools in
Solaris 11.
(Reported by Alexander Traud)
* ASTERISK-27752 - Ten seconds of silence after mp3 playback
(Reported by Sam Wierema)
* ASTERISK-27910 - [patch] res_rtp_asterisk: Allow OpenSSL
configured with no-deprecated.
(Reported by Alexander
Traud)
* ASTERISK-27906 - [patch] res_crypto: Allow OpenSSL configured
with no-deprecated.
(Reported by Alexander Traud)
* ASTERISK-27877 - app_confbridge: Add talking indicator for
ConfBridgeList AMI response
(Reported by William McCall)
* ASTERISK-27873 - documentation: Error on wiki description of
Asterisk 13 "MeetmeMute" event
(Reported by Alessandro
Polidori)
* ASTERISK-27846 - ast_coredumper: Fix OUTPUT directory
(Reported by Ted G)
* ASTERISK-27867 - [patch] libasteriskssl: Allow OpenSSL 1.0.2
configured with no-deprecated.
(Reported by Alexander
Traud)
* ASTERISK-27796 - res_hep: Allow create_address to resolve a
provided hostname
(Reported by Sebastian Gutierrez)
* ASTERISK-27820 - [patch] Add DragonFly BSD.
(Reported
by Alexander Traud)
* ASTERISK-25129 - wrong automatic ras address assignment if
multihomed
(Reported by Dmitry Melekhov)
* ASTERISK-27793 - cppcheck identifies redundant "if"
(Reported by Ilya Shipitsin)
* ASTERISK-27697 - Enable in-dialog NOTIFY on chan_pjsip
channels
(Reported by Nathan Bruning)
* ASTERISK-27770 - [patch] install_prereq: Add Slackware
(somehow).
(Reported by Alexander Traud)
* ASTERISK-27769 - [patch] install_prereq: Add Gentoo Linux.
(Reported by Alexander Traud)
* ASTERISK-27738 - [patch] install_prereq: Add Arch Linux.
(Reported by Alexander Traud)
* ASTERISK-27736 - [patch] install_prereq: Add SUSE.
(Reported by Alexander Traud)
* ASTERISK-27253 - [patch] libsrtp-2.1.x support
(Reported by Alexander Traud)
* ASTERISK-27728 - [patch] BuildSystem: Add NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27730 - PJSIP: Update bundled PJPROJECT to version
2.7.2
(Reported by Richard Mudgett)
* ASTERISK-27729 - [patch] install_prereq: Add NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27683 - [patch] BuildSystem: Allow newer autotools
on OpenBSD.
(Reported by Alexander Traud)
* ASTERISK-27348 - [patch]contrib/scripts: add a way to migrate
from chan_sip to chan_pjsip realtime
(Reported by Torrey
Searle)
* ASTERISK-27661 - Add new AMI Event for Load, Unload
(Reported by sungtae kim)
* ASTERISK-27651 - app_confbridge: Add Muted to ConfbridgeJoin
and channel snapshot headers to ConfbridgeList AMI events
(Reported by Richard Mudgett)
* ASTERISK-27647 - app_confbridge/bridge_softmix: When channel
muted report talking stopped if was talking.
(Reported by
Richard Mudgett)
* ASTERISK-27084 - Reduce verbosity while loading PBX
extensions.
(Reported by Ludovic Gasc (Eyepea))
* ASTERISK-24372 - [patch] Add config option to play a prompt
to the "winner" in app_followme
(Reported by Graham
Mainwaring)
* ASTERISK-27537 - res_pjsip: Add new AMI Action for
PJSIPShowAors
(Reported by sungtae kim)
* ASTERISK-27483 - Allow wrapuptime to be set for each queue
member
(Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-24297 - cdr.c: Minor code optimizations.
(Reported by Richard Mudgett)
* ASTERISK-27470 - Add new object for VoicemailUserEntry
(Reported by sungtae kim)
* ASTERISK-27461 - 3PCC patch for AMI "SIPnotify"
(Reported by Yasuhiko Kamata)
* ASTERISK-27449 - [PATCH] When failing to acquire target
during attended transfer, display wanted extension
(Reported by Niklas Larsson)
* ASTERISK-27456 - app_voicemail: Add new object for
VoicemailUserEntry
(Reported by sungtae kim)
* ASTERISK-27380 - ast_coredumper: allow pointing out the
asterisk binary explicitly
(Reported by Tzafrir Cohen)
* ASTERISK-23556 - Compilation warning for invert.c (array
subscript is above array bounds)
(Reported by Marcello
Ceschia)
* ASTERISK-27359 - pjproject bundled: Don't disable assertions
when --enable-dev-mode is used.
(Reported by Corey
Farrell)
* ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
(Reported by Richard Mudgett)
* ASTERISK-27335 - CDR performance needs improvement.
(Reported by Richard Mudgett)
* ASTERISK-27278 - [patch] chan_sip: Provide access to read the
full SIP Request-URI from INVITE
(Reported by David J.
Pryke)
* ASTERISK-27255 - alembic: Add support for Microsoft SQL
server
(Reported by Florian Floimair)
* ASTERISK-27220 - Enable CHANNEL function to get from and to
tag from SIP Headers
(Reported by Andre Nazario)
* ASTERISK-27169 - Google OAuth 2.0 support for XMPP / Motif
(Reported by Andrey)
* ASTERISK-27173 - Support for GMIME 3.0
(Reported by
Tzafrir Cohen)
* ASTERISK-27085 - [patch] chan_pjsip: Port SIPDtmfMode to
chan_pjsip
(Reported by Torrey Searle)
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0-rc1
Thank you for your continued support of Asterisk!
release candidate of Asterisk 16.0.0.
This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.0.0-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-27807 - iostreams: Potential DoS when client
connection closed prematurely
(Reported by Sean Bright)
* ASTERISK-27818 - Username bruteforce is possible when using
ACL with PJSIP
(Reported by John)
* ASTERISK-27658 - WebSocket frames with 0 sized payload causes
DoS
(Reported by Sean Bright)
* ASTERISK-27583 - Segmentation fault occurs in asterisk with
an invalid SDP fmtp attribute
(Reported by Sandro Gauci)
* ASTERISK-27582 - Segmentation fault occurs in Asterisk with
an invalid SDP media format description
(Reported by
Sandro Gauci)
* ASTERISK-27618 - Crash occurs when sending a repeated number
of INVITE messages over TCP or TLS transport
(Reported by
Sandro Gauci)
* ASTERISK-27640 - SUBSCRIBE message with a large Accept value
causes stack corruption
(Reported by Sandro Gauci)
New Features made in this release:
-----------------------------------
* ASTERISK-27286 - Add the ability to read the media file type
from HTTP header for playback
(Reported by Gaurav Khurana)
* ASTERISK-27704 - Add cache_pools debug option to
pjproject.conf
(Reported by Richard Mudgett)
* ASTERISK-27581 - Add new AMI Action for PJSIPShowContacts
(Reported by sungtae kim)
* ASTERISK-27547 - res_pjsip: Add new AMI Action for
PJSIPShowAuths
(Reported by sungtae kim)
* ASTERISK-27117 - core: Add support for timelen parsing to
ast_parse_arg and ACO.
(Reported by Corey Farrell)
* ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
incoming INVITE Request-URI.
(Reported by Richard Mudgett)
* ASTERISK-27413 - Add cache_media_frames debugging option.
(Reported by Richard Mudgett)
* ASTERISK-27206 - res_pjsip: No mechanism exists to limit
endpoint identification to IP only
(Reported by Ben
Merrills)
* ASTERISK-27215 - [patch]AMI : Add CancelAtxfer Action
(Reported by Thomas Sevestre)
* ASTERISK-27322 - [New Feature] Add mute and DTMF passthrough
to ARI add channel to bridge
(Reported by Darren Sessions)
* ASTERISK-27162 - [patch]chan_sip: Access incoming SIP REFER
headers in the dialplan
(Reported by Kirill Katsnelson)
* ASTERISK-27163 - chan_sip: Dialplan function SIP_HEADERS() to
complement SIP_HEADER().
(Reported by Kirill Katsnelson)
* ASTERISK-27063 - Add support for systemd socket activation
(Reported by Corey Farrell)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27978 - res_pjsip: Change default transport
keepalive to preserve behavior
(Reported by Joshua Colp)
* ASTERISK-27880 - [patch] pjproject_bundled: Repair
./configure --with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27810 - BASIC-RETRANS: Implement receive
(Reported by Benjamin Keith Ford)
* ASTERISK-27972 - res_sorcery_config: Allow object name based
matching
(Reported by Joshua Colp)
* ASTERISK-27965 - module: Remove old modules, update support
levels
(Reported by Joshua Colp)
* ASTERISK-25548 - stasis: Improve message type "Use of before
init/after destruction" error
(Reported by Joshua Colp)
* ASTERISK-27967 - srtp: rejecting short sdes lifetimes
incompatible with obihai ATAs
(Reported by Nick French)
* ASTERISK-27961 - res_pjsip: Spurious ERROR logging when
printing headers in sip_msg
(Reported by Nick French)
* ASTERISK-27563 - pjsip modules always get -O2 even when
DONT_OPTIMIZE is set
(Reported by George Joseph)
* ASTERISK-27347 - [patch] pjproject_bundled: Disable TCP/TLS
keep-alives.
(Reported by Alexander Traud)
* ASTERISK-27957 - PJSIP proposes ICE candidates on answer even
if not in offer
(Reported by Torrey Searle)
* ASTERISK-27938 - [patch] Compile fails with `IPTOS_MINCOST'
undeclared.
(Reported by Alexander Traud)
* ASTERISK-27955 - res_pjsip_session: sdp group:BUNDLE
attribute truncated
(Reported by Kevin Harwell)
* ASTERISK-27956 - res_pjsip_pubsub: segfault in function
publish_expire
(Reported by Alexei Gradinari)
* ASTERISK-27949 - res_pjsip_rfc3326: A lot of endpoints do not
correctly handle two Reason headers
(Reported by Ross
Beer)
* ASTERISK-27763 - res_pjsip_session: Initial INVITE with
audio+fax results in 488 instead of declining stream
(Reported by Thiago Coutinho)
* ASTERISK-27657 - res_pjsip_t38: ATA fails with hangupcause
58(Bearer capability not available)
(Reported by Jared
Hull)
* ASTERISK-27080 - res_pjsip_t38: Slow T.38 re-invite rejection
if remote leg has T.38 disabled
(Reported by Torrey
Searle)
* ASTERISK-26686 - res_pjsip: Lock inversion in transport
management
(Reported by Ross Beer)
* ASTERISK-27939 - [patch] bridge_softmix_binaural: Enable
FFTW3 in Solaris 11.
(Reported by Alexander Traud)
* ASTERISK-27783 - res_pjsip_pubsub: apparent crash on
shutdown
(Reported by Kevin Harwell)
* ASTERISK-27870 - app_confbridge: Conference bridge and
announcer channels are not removed if conference is ended as
soon as it starts
(Reported by Robert Mordec)
* ASTERISK-27909 - cdr: Deadlock with submit_scheduled_batch
and submit_unscheduled_batch
(Reported by Denis Lebedev)
* ASTERISK-26987 - pbx_dundi: Asterisk crashes when unloading
module pbx_dundi.so with dundi peers
(Reported by Kirsty
Tyerman)
* ASTERISK-27943 - AMI: Action SendText needs to use the
correct thread.
(Reported by Richard Mudgett)
* ASTERISK-27942 - res_pjsip_messaging doesn't accept
application/* content-types.
(Reported by George Joseph)
* ASTERISK-27936 - res_pjsip_session doesn't update media when
a 200 comes in with a different port than a 183
(Reported
by George Joseph)
* ASTERISK-27933 - [patch] uuid: Enable UUID in Solaris 11.
(Reported by Alexander Traud)
* ASTERISK-27625 - channels: CHECK_BLOCKING is ineffective
(Reported by Corey Farrell)
* ASTERISK-27931 - [patch] BuildSystem: Enable ./configure in
Solaris 11.
(Reported by Alexander Traud)
* ASTERISK-27926 - [patch] bootstrap.sh: find -maxdepth is not
POSIX compatible.
(Reported by Alexander Traud)
* ASTERISK-27903 - menuselect: GCC 8: restrict-qualified
parameter passed and aliased.
(Reported by Alexander
Traud)
* ASTERISK-27914 - [patch] tests/test_utils: Repair ./configure
--with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27705 - chan_iax2: Stops listening for traffic
(Reported by Kirsty Tyerman)
* ASTERISK-27848 - rtp: DTMF Breaks With telephony-event/16000
(Reported by Dominic)
* ASTERISK-27908 - [patch] crypto.h: Repair ./configure
--with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27905 - [patch] res_srtp: Repair ./configure
--with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27888 - SQL fetch error on query which return 0
columns
(Reported by Alexei Gradinari)
* ASTERISK-27902 - chan_pjsip isn't updating hangupcause on 4XX
responses
(Reported by George Joseph)
* ASTERISK-27901 - [patch] ooh323c: GCC 8: output truncated
before terminating nul.
(Reported by Alexander Traud)
* ASTERISK-27872 - res_pjsip: Modified qualify_frequency
doesn't effect until pjsip reload
(Reported by Alexei
Gradinari)
* ASTERISK-27094 - res_fax: Deadlock when using Local channels
and fax gateway
(Reported by David Brillert)
* ASTERISK-25261 - Manager events for MeetMe have incorrectly
documented key name 'Usernum' - should be 'User'
(Reported
by Francois Blackburn)
* ASTERISK-27878 - [patch] tcptls.h: Repair ./configure
--with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27876 - [patch] tcptls: Allow OpenSSL configured
with no-dh.
(Reported by Alexander Traud)
* ASTERISK-27874 - [patch] tcptls: Allow OpenSSL 1.1.x
configured with enable-ssl3-method no-deprecated.
(Reported by Alexander Traud)
* ASTERISK-27845 - Codec-Change Re-INVITE during DTMF can cause
marker bit error
(Reported by Torrey Searle)
* ASTERISK-27831 - res_rtp_asterisk: Add support for
abs-send-time RTP extension
(Reported by Joshua Colp)
* ASTERISK-27863 - config/ast_destroy_realtime_fields:
successful DELETE is treated as failed
(Reported by Alexei
Gradinari)
* ASTERISK-27865 - [patch]: tcptls: Repair ./configure
--with-ssl=PATH.
(Reported by Alexander Traud)
* ASTERISK-27760 - Asterisk ODBC Voicemail Prompt storage fails
with recent MariaDB version.
(Reported by Nic Colledge)
* ASTERISK-27853 - Incorrect error reported when
leaving/retrieving a ODBC voicemail
(Reported by Nic
Colledge)
* ASTERISK-27726 - chan_mobile: presents incorrect inbound
Caller-ID names
(Reported by Brian)
* ASTERISK-27861 - [patch] res_pjsip_endpoint_identifier_ip:
Unregister the module for headers.
(Reported by Alexander
Traud)
* ASTERISK-27852 - cli: "manager show settings" mislabels HTTP
timeout as being minutes.
(Reported by Corey Farrell)
* ASTERISK-27824 - Fix issues exposed by GCC 8
(Reported
by George Joseph)
* ASTERISK-27850 - [patch] rtp_engine: Allow Media Formats with
add_static_payload(-1) on egress again.
(Reported by
Alexander Traud)
* ASTERISK-27811 - [patch] sip_to_pjsip: Enable python3
compatibility.
(Reported by Alexander Traud)
* ASTERISK-27841 - digest over for manager (ami) over http
fails on too long uris
(Reported by Jaco Kroon)
* ASTERISK-26570 - Macro allows an infinite loop of dialplan
inclusion resulting in a crash
(Reported by Tzafrir Cohen)
* ASTERISK-27572 - cdr_mysql creates empty records if
reconnects when mysql was not up on module load
(Reported
by Tzafrir Cohen)
* ASTERISK-27801 - Asterisk got stuck while enabling "ari set
debug all on"
(Reported by shaurya jain)
* ASTERISK-27795 - chan_sip: one way / no audio with srtp
(Reported by Florian Kaiser)
* ASTERISK-27800 - One way audio when calling from Asterisk(sip
trunk) to another number where both are connected to a SBC using
TLS+SRTP
(Reported by Artur Pires)
* ASTERISK-26806 - pjsip_options: rework to make more
efficient
(Reported by Kevin Harwell)
* ASTERISK-27814 - translate: interpolated frames are not
passed through
(Reported by Kevin Harwell)
* ASTERISK-27812 - When the ooh323 debug is on there is no
ringing signal to incoming calls via H323 trunk.
(Reported
by Dimos)
* ASTERISK-26893 - No "alert" or "progress" in chan_ooh323 if
debug is enabled only on the module
(Reported by Marco
Giordani)
* ASTERISK-27804 - bridge_softmix / app_confbridge: Add support
for combining REMB reports
(Reported by Joshua Colp)
* ASTERISK-27639 - [patch] BuildSystem: Enable IMAP storage on
FreeBSD and DragonFly BSD.
(Reported by Alexander Traud)
* ASTERISK-27418 - app_confbridge: "core show profile bridge"
does not output "sfu" when video_mode is sfu
(Reported by
Carlos Chavez)
* ASTERISK-27809 - [patch] utils/pval: Add -lBlocksRuntime for
compiler clang conditionally.
(Reported by Alexander
Traud)
* ASTERISK-27808 - [patch] chan_vpb: Avoid GNU old-style field
designator extension.
(Reported by Alexander Traud)
* ASTERISK-27806 - BASIC-RETRANS: Implement send
(Reported by Benjamin Keith Ford)
* ASTERISK-27774 - res_musiconhold: Music on hold restarts
after every announcement
(Reported by lvl)
* ASTERISK-27782 - cdr_mysql: Missing MYSQL_PORT definition
(Reported by Evandro César Arruda)
* ASTERISK-27614 - res_pjsip_session: SDP origin does not use
resolved address
(Reported by John M.)
* ASTERISK-27776 - res_rtp_asterisk: Add support for sending
RTCP feedback messages
(Reported by Joshua Colp)
* ASTERISK-27740 - chan_sip: New Channel creation from new SIP
dialog with Replaces failed to be properly tracked and
destroyed
(Reported by Shannon Price)
* ASTERISK-27786 - app_confbridge: Add ability to enable and
configure REMB support
(Reported by Joshua Colp)
* ASTERISK-27706 - PJSIP: Deadlock shutting down subscription
TCP connection and sending subscription message.
(Reported
by Ross Beer)
* ASTERISK-27688 - res_pjsip: Crash on TCP PJSIP Transport
Disconnect
(Reported by Ross Beer)
* ASTERISK-27758 - res_rtp_asterisk: Add support for raising
RTCP feedback messages
(Reported by Joshua Colp)
* ASTERISK-26366 - rtp: RTCP messages with REMB trigger fast
picture update
(Reported by Joshua Colp)
* ASTERISK-27773 - Command line not being parsed correctly with
getopt not from glibc
(Reported by Guido Falsi)
* ASTERISK-27435 - [patch] configure:
pjsip_evsub_set_uas_timeout not found.
(Reported by
Alexander Traud)
* ASTERISK-27761 - [patch] BuildSystem: With external editline,
do not require libs for internal editline.
(Reported by
Alexander Traud)
* ASTERISK-27755 - ConfBridge: raise ConfbridgeTalking when put
on hold and clear talking status
(Reported by Kevin
Harwell)
* ASTERISK-27743 - Generic PLC doesn't work if the 2 codecs on
a channel are equal
(Reported by George Joseph)
* ASTERISK-27745 - [patch] BuildSystem: Remove unused
dependency on libltdl.
(Reported by Alexander Traud)
* ASTERISK-12841 - [patch] Make format_ogg_vorbis work on
OpenBSD
(Reported by Michiel van Baak)
* ASTERISK-27720 - [patch] BuildSystem: Enable Advanced Linux
Sound Architecture (ALSA) in NetBSD.
(Reported by
Alexander Traud)
* ASTERISK-27741 - res_pjsip_rfc3326.c
rfc3326_use_reason_header doesn't account for more than one
'Reason' header
(Reported by Ross Beer)
* ASTERISK-27734 - [patch] BuildSystem: Enable IMAP storage on
openSUSE and Arch Linux.
(Reported by Alexander Traud)
* ASTERISK-27686 - [patch] install_prereq: Update FreeBSD
libraries.
(Reported by Alexander Traud)
* ASTERISK-27733 - [patch] res_srtp: Add support for libsrtp2.x
on openSUSE.
(Reported by Alexander Traud)
* ASTERISK-11015 - NetBSD Build Needs RPATH set in 1.2.25
(Reported by Curt Sampson)
* ASTERISK-27641 - BuildSystem: Enable Better Backtraces in
FreeBSD.
(Reported by Alexander Traud)
* ASTERISK-27671 - Deprecate legacy modules
(Reported by
Corey Farrell)
* ASTERISK-25586 - uuid_generate_random detection failure
(Reported by John Nemeth)
* ASTERISK-27721 - [patch] BuildSystem: Enable PortAudio in
NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27715 - [patch] BuildSystem: AC_PATH_PROG sets to
colon character when not found.
(Reported by Alexander
Traud)
* ASTERISK-27554 - res_pjsip_rfc3326: Order of 'Reason' headers
break many endpoints
(Reported by Ross Beer)
* ASTERISK-27703 - AMI Action VoicemailUsersList returns 0
MessageCount
(Reported by Sébastien Duthil)
* ASTERISK-27674 - chan_sip: RTP framing issues on outgoing
calls
(Reported by Jean Aunis - Prescom)
* ASTERISK-27441 - PJSIP: Forked INVITE SDP negotiation gets
one way audio.
(Reported by lvl)
* ASTERISK-27718 - [patch] BuildSystem: Enable Lua in NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27722 - [patch] BuildSystem: Depend not implicitly
but explicitly on external libraries.
(Reported by
Alexander Traud)
* ASTERISK-27719 - [patch] res_http_post: Enable GMime in
NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27716 - [patch] BuildSystem: Enable autotools in
NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27714 - [patch] chan_unistim: NetBSD has an
incompatible struct in_pktinfo.
(Reported by Alexander
Traud)
* ASTERISK-27713 - [patch] BuildSystem: Cast any intptr_t
explicitly to its proposed type.
(Reported by Alexander
Traud)
* ASTERISK-27712 - [patch] BuildSystem: Detect whether
uselocale(.) is available.
(Reported by Alexander Traud)
* ASTERISK-27711 - [patch] BuildSystem: Avoid re-defining of
pthread_* on NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27710 - [patch] BuildSystem: Install init scripts on
openSUSE Tumbleweed.
(Reported by Alexander Traud)
* ASTERISK-27709 - [patch] BuildSystem: Avoid == for comparison
in ./configure.
(Reported by Alexander Traud)
* ASTERISK-27610 - app_amd.so returning TOOLONG before reaching
the timeout
(Reported by Michael Cargile)
* ASTERISK-26688 - Documentation: voicemail.conf.sample shows
512 limit for emailbody field, however this is only true if
compiled with LOW_MEMORY option
(Reported by Fran Vicente)
* ASTERISK-27568 - PJSIP: Crash during SIP attended transfer.
(Reported by Bryan Walters)
* ASTERISK-27659 - Output from rawman truncated if output is
long enough
(Reported by Bojan NemÄiÄ)
* ASTERISK-27692 - bridging: Sometimes cloning the stream
topology causes a crash
(Reported by Richard Mudgett)
* ASTERISK-27488 - core: If frame with unnegotiated format is
read crash will occur
(Reported by Sébastien Duthil)
* ASTERISK-24488 - Wrong remote identity and target in dialog
package XML in NOTIFY
(Reported by Alejandro Padilla)
* ASTERISK-24386 - Asterisk "doc/lang/language-criteria.txt"
needs update or removal.
(Reported by Rusty Newton)
* ASTERISK-27646 - ICE fails with no candidate nominated
(Reported by Thomas Guebels)
* ASTERISK-27689 - [patch] rtp_engine: Load format name / mime
type in uppercase again.
(Reported by Alexander Traud)
* ASTERISK-27679 - res_pjsip: Endpoint destruction does not
free DTLS configuration
(Reported by Mak Dee)
* ASTERISK-27684 - [patch] install_prereq: Update OpenBSD
libraries.
(Reported by Alexander Traud)
* ASTERISK-27680 - [patch] res_calendar: Specialized calendars
depend on symbols of general calendar.
(Reported by
Alexander Traud)
* ASTERISK-27681 - [patch] BuildSystem: Enable IMAP storage on
OpenBSD.
(Reported by Alexander Traud)
* ASTERISK-27677 - [patch] BuildSystem: Enable system provided
libedit on OpenBSD.
(Reported by Alexander Traud)
* ASTERISK-27670 - [patch] BuildSystem: Remove chan_h323
leftovers.
(Reported by Alexander Traud)
* ASTERISK-27595 - [patch] BuildSystem: Invoke ldconfig with
previous paths.
(Reported by Alexander Traud)
* ASTERISK-27631 - [patch] BuildSystem: Do not warn when bash
is not installed.
(Reported by Alexander Traud)
* ASTERISK-27666 - chan_sip: Crash processing CANCEL request
(Reported by Leandro Dardini)
* ASTERISK-27584 - Internal pjproject build doesn't disable
bcg729
(Reported by Stuart Henderson)
* ASTERISK-27669 - [patch] codecs: Add support for WebRTC iLBC
2.0.
(Reported by Alexander Traud)
* ASTERISK-27634 - Determine if the internal editline and
stdtime libraries are still relevant
(Reported by George
Joseph)
* ASTERISK-27642 - [patch] backtrace: Avoid
-Wlogical-not-parentheses.
(Reported by Alexander Traud)
* ASTERISK-27555 - [patch] install_prereq: Update Debian/Ubuntu
libraries.
(Reported by Alexander Traud)
* ASTERISK-27656 - CDR: Leaking channel snapshots allocated by
stasis_channel.c
(Reported by Kristijan Vrban)
* ASTERISK-27426 - chan_console: cannot read and write at the
same time with alsa backend
(Reported by Tzafrir Cohen)
* ASTERISK-27621 - (null) string tailing after AsyncAGIEnd AMI
event
(Reported by sungtae kim)
* ASTERISK-27652 - Null pointer Crash in PJSIP MWI
(Reported by Joshua Elson)
* ASTERISK-27571 - res_pjsip: If SIP response is received
during shutdown a crash may occur
(Reported by Joshua
Colp)
* ASTERISK-27619 - Build System: Require compiler to provide
built-in support for atomic references.
(Reported by Corey
Farrell)
* ASTERISK-27612 - Subscriptions Persist After Expiration and
TCP/TLS Disconnect
(Reported by Ross Beer)
* ASTERISK-27637 - [patch] BuildSystem: Enable autotools in
FreeBSD.
(Reported by Alexander Traud)
* ASTERISK-27635 - [patch] app_voicemail: Avoid always true
warnings with clang.
(Reported by Alexander Traud)
* ASTERISK-27599 - [patch] install_prereq: Update
RHEL/CentOS/Fedora libraries.
(Reported by Alexander
Traud)
* ASTERISK-26563 - core: macOS devmode build fails: variable
'freeswap' set but not used
(Reported by David M. Lee)
* ASTERISK-27630 - [patch] editline: Avoid shifting a negative
signed value.
(Reported by Alexander Traud)
* ASTERISK-16172 - Problems with siren14 codec; problems with
siren7 sound files.
(Reported by Steve Murphy)
* ASTERISK-16951 - [patch] configure.ac in 1.4.37 broken with
autoconf 2.60
(Reported by Stéphan Kochen)
* ASTERISK-27603 - [patch] install_prereq: Download latest
Jansson.
(Reported by Alexander Traud)
* ASTERISK-27620 - New module loader aborts startup if a
required module declines load.
(Reported by snuffy)
* ASTERISK-27607 - [patch] res_config_mysql: Avoid the header
mysql_version.h.
(Reported by Alexander Traud)
* ASTERISK-24598 - When running
./contrib/scripts/install_prereq install-unpackaged pjproject is
installed in wrong place
(Reported by PowerPBX)
* ASTERISK-27602 - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs
a directory.
(Reported by Alexander Traud)
* ASTERISK-27600 - [patch] BuildSystem: Allow make clean all
again.
(Reported by Alexander Traud)
* ASTERISK-27598 - [patch] install_prereq: Support package
manager DNF.
(Reported by Alexander Traud)
* ASTERISK-26596 - Placing call on hold temporarily locks up
set
(Reported by Igor Goncharovsky)
* ASTERISK-27596 - [patch] BuildSystem: Use the detected name
for MD5 everywhere.
(Reported by Alexander Traud)
* ASTERISK-27594 - [patch] BuildSystem: Invoke install not in
GNU but POSIX style.
(Reported by Alexander Traud)
* ASTERISK-27593 - [patch] BuildSystem: In OpenBSD, xmlstarlet
is xml.
(Reported by Alexander Traud)
* ASTERISK-27592 - [patch] BuildSystem: Detect external library
Lua in version 5.3.
(Reported by Alexander Traud)
* ASTERISK-27491 - res_pjsip_endpoint_identifier_ip only
matches against header if match by ip fails
(Reported by
George Joseph)
* ASTERISK-26832 - res_pjsip: Segfault when calling
pjsip_hdr_print_on in sip_msg.c:581
(Reported by Ross
Beer)
* ASTERISK-27589 - [patch] BuildSystem: Avoid $EUID and use id
-u instead.
(Reported by Alexander Traud)
* ASTERISK-27585 - [patch] BuildSystem: Resolve resolv.h not
via Generic but Particular Header-Check.
(Reported by
Alexander Traud)
* ASTERISK-27575 - menuselect : remove obsolete TRACE_FRAMES
compiler flag
(Reported by Jean Aunis - Prescom)
* ASTERISK-27576 - [patch] res_config_pgsql: Avoid typecasting
an int to unsigned char.
(Reported by Alexander Traud)
* ASTERISK-27560 - [patch] clang 5 does not know
-Wno-format-truncation
(Reported by Alexander Traud)
* ASTERISK-27578 - [patch] app_osplookup.c: Avoid a format
truncation.
(Reported by Alexander Traud)
* ASTERISK-27577 - [patch] chan_ooh323: Avoid typecasting an
int to unsigned short.
(Reported by Alexander Traud)
* ASTERISK-27534 - chan_sip: Assumes iostream is non-NULL when
it may not be
(Reported by Lubos Dolezel)
* ASTERISK-27549 - [patch] translate: Avoid absolute value on
unsigned substraction.
(Reported by Alexander Traud)
* ASTERISK-27566 - res_pjsip_session: Improve WebRTC interop
with bundling during renegotiation
(Reported by Joshua
Colp)
* ASTERISK-27553 - [patch] res_curl: Avoid error message on
unload.
(Reported by Alexander Traud)
* ASTERISK-27557 - [patch] clang 5.0: implicit conversion to
char changes value to negative.
(Reported by Alexander
Traud)
* ASTERISK-27550 - [patch] bridge_softmix: Avoid warning about
an uninitialized variable.
(Reported by Alexander Traud)
* ASTERISK-27559 - [patch] editline: Avoid comparison between
pointer and zero character constant.
(Reported by
Alexander Traud)
* ASTERISK-27558 - [patch] codec_gsm: Avoid shifting a negative
signed value.
(Reported by Alexander Traud)
* ASTERISK-25329 - Asterisk configure fails on 'cannot find
ptlib-config', despite ptlib-config existing
(Reported by
Rusty Newton)
* ASTERISK-27552 - [patch] chan_ooh323: Limit outgoinglimit to
positive values as intended.
(Reported by Alexander Traud)
* ASTERISK-27551 - [patch] ooh323cDriver: Fix typo in header
guard.
(Reported by Alexander Traud)
* ASTERISK-26046 - [patch] Avoid obsolete warnings on
autoconf.
(Reported by Alexander Traud)
* ASTERISK-20346 - Modules need to ensure that any functions,
apps, AMI actions, etc. they register are unregistered if the
module declines loading
(Reported by Mark Michelson)
* ASTERISK-27539 - 'cdr submit' fails: batch mode not enabled.
(Reported by Tzafrir Cohen)
* ASTERISK-27498 - ICE candidate parser - ICE foundation
parsing too short
(Reported by Michele Prà )
* ASTERISK-25128 - Datastore: Implement automatic module
references.
(Reported by Corey Farrell)
* ASTERISK-27366 - Asterisk Turkish Language Set Problem
(Reported by Halil Ä°brahim YILDIZ)
* ASTERISK-23133 - Documentation fix - MASTER_CHANNEL
Unexpected Behaviour
(Reported by Shane Mitchell)
* ASTERISK-27531 - Compiler optimizations can break module load
sequence.
(Reported by abelbeck)
* ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
Contact crashes asterisk
(Reported by Ross Beer)
* ASTERISK-24198 - Typo's
(Reported by Walter Doekes)
* ASTERISK-27229 - bridge: Old channel video source not set to
NULL after unref
(Reported by Richard Kenner)
* ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
(Reported by Corey Farrell)
* ASTERISK-25079 - AMI bridge of channels results in MOH not
destroyed and robotic audio on one channel
(Reported by
Zane Conkle)
* ASTERISK-27490 - chan_console: 'set active' fails to work
(Reported by Tzafrir Cohen)
* ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
read()
(Reported by Abhay Gupta)
* ASTERISK-24756 - ConfBridge sound_muted does not work from
CLI or AMI
(Reported by Thomas Frederiksen)
* ASTERISK-25649 - Transfer application does not work with
Local channels - documentation misleading
(Reported by
Ivan Ullmann)
* ASTERISK-25869 - chan_sip: "rejected because extension not
found" should be logged as a security event
(Reported by
Brian J. Murrell)
* ASTERISK-27440 - Strictrtp has issues to qualify video rtp
streams
(Reported by Wim De Vlaminck)
* ASTERISK-19657 - Coverity Report: Fix issues for error type
CHAR_IO
(Reported by Matt Jordan)
* ASTERISK-27175 - iax.conf demo peer is invalid
(Reported by Tzafrir Cohen)
* ASTERISK-27430 - README refers to security documents that do
not exist.
(Reported by Corey Farrell)
* ASTERISK-20281 - "core set verbose" behaves strangely, can't
alias it, cli.conf example broken
(Reported by Tim
Ringenbach at Asteria Solutions Group)
* ASTERISK-27382 - crash after an invalid rtcp packet from GT48
FXS gateway
(Reported by Tzafrir Cohen)
* ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
RTCP packet will write past where it should
(Reported by
Vitezslav Novy)
* ASTERISK-27408 - Identify causes and fix
pjsip/resolver/srv/failover/in_dialog/transport_tcp
(Reported by Corey Farrell)
* ASTERISK-18411 - Queue members with hints for state_interface
get stuck in "In Use" state.
(Reported by Steven T.
Wheeler)
* ASTERISK-26131 - chan_sip: Crash Asterisk (in
sip_request_call at chan_sip.c) by making a call to a single
character in a dot pattern match
(Reported by Dwayne
Hubbard)
* ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
not applied on reload
(Reported by John Bigelow)
* ASTERISK-27460 - CDR: Deadlock using AMI Originate with
Variable CDR(amaflags)=...
(Reported by Richard Mudgett)
* ASTERISK-27453 - RTP: Blind transfer direct media scenario
results in one way audio.
(Reported by Richard Mudgett)
* ASTERISK-20643 - SIP ICE support - remove hardcoded
limitation on SDP size, make ICE support disabled by default in
SIP, maybe provide a better warning message
(Reported by
Roy)
* ASTERISK-27457 - chan_sip: Guests disallowed via TCP (or TLS)
if existing peer from same IP.
(Reported by Alexander
Traud)
* ASTERISK-26980 - pjsip: Clean up WebRTC disables
(Reported by abelbeck)
* ASTERISK-27452 - Security: chan_skinny: Memory exhaustion if
flooded with unauthenticated requests
(Reported by George
Joseph)
* ASTERISK-27454 - res_http_post: Don't require
GMIME_MAJOR_VERSION
(Reported by Joshua Colp)
* ASTERISK-23735 - Transcoding makes bad choice in high-rate
translations
(Reported by Richard Kenner)
* ASTERISK-27445 - ARI: Updating a bridge gives wrong error
message.
(Reported by Frank Durden)
* ASTERISK-24662 - [patch] column and row headers for Signed
Linear format variants in output of 'core show translation' are
ambiguous
(Reported by Rusty Newton)
* ASTERISK-27353 - H323 audio starts with a delay of 2
seconds.
(Reported by Marco Giordani)
* ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
media
(Reported by Kevin Harwell)
* ASTERISK-27437 - [patch] ICE: server-reflexive candidates
(srflx) with Dual-Stack.
(Reported by Alexander Traud)
* ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
IPv6 addresses.
(Reported by Alexander Traud)
* ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
(Reported by Ivan Larionov)
* ASTERISK-27431 - Asterisk fails to build when openssl headers
are not installed.
(Reported by Corey Farrell)
* ASTERISK-27421 - RTP source learning not working with devices
that have some clock issues
(Reported by nappsoft)
* ASTERISK-27361 - Attended transfer crashes in Asterisk
13.17.2
(Reported by Alessandro Pimenta)
* ASTERISK-27238 - Bridging: Crash freeing a frame that's
already been freed
(Reported by Richard Kenner)
* ASTERISK-27412 - core: Audiohook freeing interpolated frame
when it shouldn't.
(Reported by Mikhail)
* ASTERISK-27423 - app_record: We set the RECORD_STATUS
channel variable before closing the file
(Reported by
George Joseph)
* ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
insert same ip address in "source ip address" and "destination
ip address" fields in HEP packets
(Reported by Max Norba)
* ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
is equal to RemoteAddress)
(Reported by Vasilii Rogin)
* ASTERISK-27415 - asterisk.conf: Setting astctl without
setting astrundir is ineffective.
(Reported by Corey
Farrell)
* ASTERISK-27411 - pjsip: TCP connections may not be destroyed
(Reported by Joshua Colp)
* ASTERISK-27404 - DEBUG_FD_LEAKS does not record socketpair,
timerfd_create or eventfd.
(Reported by Corey Farrell)
* ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
responses.
(Reported by Corey Farrell)
* ASTERISK-27337 - chan_sip: Security vulnerability with client
code header (revisited)
(Reported by Richard Mudgett)
* ASTERISK-27319 - (Security) Function in PJSIP 2.7
miscalculates the length of an unsigned long variable in 64bit
machines
(Reported by Kim youngsung)
* ASTERISK-27391 - Regression: Deadlock between AOR named lock
and pjproject grp lock
(Reported by shaurya jain)
* ASTERISK-27393 - res_pjsip: Crash occurs when an empty
contact read from astdb or database
(Reported by Aaron An)
* ASTERISK-27290 - res_pjsip: PIDF contact field has
malformed/invalid XML
(Reported by basildane)
* ASTERISK-27032 - res_pjsip: TLS options do not handle empty
values
(Reported by seanchann.zhou)
* ASTERISK-27395 - srtp: Add support for ephemeral DTLS
certificates
(Reported by Sean Bright)
* ASTERISK-26426 - format_ogg_opus: remove from source
(Reported by Kevin Harwell)
* ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
enabled but not configured.
(Reported by Alexander Traud)
* ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
(Reported by Alexander Traud)
* ASTERISK-27378 - Modules: Fix issues with CLI completion.
(Reported by Corey Farrell)
* ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
character isn't allowed any more
(Reported by Michael
Maier)
* ASTERISK-27364 - channel: Crash when fax gateway is in use
with PJSIP
(Reported by Jared Hull)
* ASTERISK-27390 - Audit menuselect module dependencies
(Reported by Corey Farrell)
* ASTERISK-27389 - Optional API modules should not allow
unload.
(Reported by Corey Farrell)
* ASTERISK-27369 - Bridge() dialplan application fails without
setting BRIDGERESULT channel variable
(Reported by James
Terhune)
* ASTERISK-27067 - res_ari_channels: channel_state_invalid
always leaks snapshot reference.
(Reported by Marin
Odrljin)
* ASTERISK-27379 - stream: Allow streams on a topology to be
put into groups
(Reported by Joshua Colp)
* ASTERISK-27374 - alembic: PJSIP scripts are missing column
bundle in ps_endpoints table
(Reported by Florian
Floimair)
* ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
documentation
(Reported by Igor Goncharovsky)
* ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
'imap_delete_old_greeting'
(Reported by Anthony Messina)
* ASTERISK-27194 - jitterbuffer: Does not handle case where
translator returns null frame.
(Reported by Joshua Elson)
* ASTERISK-27372 - ARI: Node ARI client broken in latest
versions of 13 and 14
(Reported by Benjamin Keith Ford)
* ASTERISK-26639 - core: Disabling xmldoc support does not
work. Also results in abort during Asterisk startup.
(Reported by Mr Dini)
* ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
absence of the Expires header field with an unsubscribe action.
(Reported by Jonathan Cloots)
* ASTERISK-25960 - The config_hook unit test causes Asterisk to
crash if run a second time
(Reported by George Joseph)
* ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
when rtp_ipv6 set to yes
(Reported by Martin Cisárik)
* ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
curl is loaded
(Reported by Ronald Raikes)
* ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
but first on SDP media level.
(Reported by Alexander
Traud)
* ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
Assertion on un/re-load: mod.id == -1
(Reported by Tzafrir
Cohen)
* ASTERISK-23462 - Cannot disable SIP debugging via CLI after
enabling with conf file option - also 'sip set debug off'
reports debugging disabled, when it really isn't
(Reported
by Rusty Newton)
* ASTERISK-27350 - app_macro deprecation
(Reported by
Corey Farrell)
* ASTERISK-27354 - bridge_softmix: When a channel leaves add in
any missing participant streams
(Reported by Joshua Colp)
* ASTERISK-27333 - sip_to_pjsip not correctly handling
disallow=all directive
(Reported by Torrey Searle)
* ASTERISK-27343 - Fails to build in FreeBSD due to
sys/sysmacros.h not existing there
(Reported by Guido
Falsi)
* ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
(o=) contains local address.
(Reported by Alexander Traud)
* ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
configured codecs, but subset based on caller
(Reported by
lvl)
* ASTERISK-27340 - backtrace.c: Crash due to double-free.
(Reported by Corey Farrell)
* ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
stopping.
(Reported by Alexander Traud)
* ASTERISK-27047 - res_pjsip: user=phone added to Anonymous
caller-id when it shouldn't be.
(Reported by dtryba)
* ASTERISK-26988 - res_pjsip_session: user_eq_phone adds double
user=phone parameters to URIs
(Reported by dtryba)
* ASTERISK-27301 - [patch] app_queue: Music On Hold for
real-time queues is not reset to default
(Reported by
Nathan Bruning)
* ASTERISK-25266 - Application Originate returns SUCCESS to
ORIGINATE_STATUS upon failure to originate
(Reported by
Allen Ford)
* ASTERISK-27270 - cdr_mysql: various crashes at second module
reload if cdr_mysql.conf is configured
(Reported by
Tzafrir Cohen)
* ASTERISK-27328 - Missing openssl dependencies in
res_rtp_asterisk and tcptls
(Reported by Tzafrir Cohen)
* ASTERISK-27192 - res_pjsip: Loss of SIP registrations causing
unavailable endpoints
(Reported by Richard Mudgett)
* ASTERISK-27305 - res_ari: Memory leaks in ARI when using
Content-Type: application/json
(Reported by David Hajek)
* ASTERISK-26922 - chan_sip: tcpbind uses wrong source address
(Reported by Ksenia)
* ASTERISK-27324 - [patch] Dual-Stack server cannot be used as
IPv4 client via TCP/TLS
(Reported by Alexander Traud)
* ASTERISK-27317 - vector: multiple evaluation of elem in
AST_VECTOR_ADD_SORTED.
(Reported by Corey Farrell)
* ASTERISK-27318 - res_pjsip_mwi: uninitialized value from
ast_strings_match
(Reported by Corey Farrell)
* ASTERISK-27284 - Status of RFC 3323 and PJSIP
(Reported
by dtryba)
* ASTERISK-27296 - [patch] False positive busy checks when
icalendar's recurrence-id mechanism is involved
(Reported
by Benoît Dereck-Tricot)
* ASTERISK-27216 - app_queue: does its
check-makeannouncement-logic twice each head-caller-loop
(Reported by Stefan Engström)
* ASTERISK-27298 - Problem with expires on pjsip /
outbound-publish
(Reported by Cyrille Demaret)
* ASTERISK-27295 - Contact is improperly translated after
d178f497
(Reported by Sean Bright)
* ASTERISK-27292 - Multiple RTP Stream Created Breaking RFC2833
(SSRC Changes)
(Reported by Ross Beer)
* ASTERISK-27289 - A codeblock that maintains a bug,but maybe
the codeblock will never run
(Reported by Huangyx)
* ASTERISK-27277 - bridge: Renegotiate if source stream
changes.
(Reported by Joshua Colp)
* ASTERISK-27264 - res_pjsip_session: Crashes after sending
PRACK and receiving 200 OK
(Reported by Daniel Heckl)
* ASTERISK-27283 - Realtime config fail with PostgreSQL version
before 9.1
(Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-27260 - [pjsip] chan_pjsip_indicate: Don't know how
to indicate condition 36
(Reported by Daniel Heckl)
* ASTERISK-27257 - bridge_native_rtp: half-way direct media
when using early bridging
(Reported by Jean Aunis -
Prescom)
* ASTERISK-16898 - SRTP unprotect: authentication failure when
RTP sequence number switches from 65535 -> 0
(Reported by
Marcello Ceschia)
* ASTERISK-27279 - Crash in pubsub_on_rx_request NULL pointer -
Possible PJSIP Vulnerability
(Reported by Ross Beer)
* ASTERISK-25524 - module reload res_calendar.so does not
reload everything in calendar.conf
(Reported by Jesper)
* ASTERISK-27274 - RTCP needs better packet validation to
resist port scans.
(Reported by Richard Mudgett)
* ASTERISK-27252 - RTP: One way audio with direct media and
strictrtp=yes.
(Reported by Richard Mudgett)
* ASTERISK-24588 - res_calendar does not process CalDAV from
Owncloud [fix included]
(Reported by Stefan Gofferje)
* ASTERISK-25523 - res_calendar: Warning about invalid channel
value (for notification) occurs even when event has no
notification configured.
(Reported by Jesper)
* ASTERISK-21399 - RTP Multicast of L16 (type 10): Asterisk and
wireshark disagree
(Reported by Tzafrir Cohen)
* ASTERISK-27248 - [patch]external_media_address and
external_signaling_address don't always honor localnet
(Reported by Walter Doekes)
* ASTERISK-27165 - CDR: CDR(start,u) function won't work in
cdr_custom config
(Reported by Jacek Konieczny)
* ASTERISK-24066 - res_smdi: convert to astobj2
(Reported
by Corey Farrell)
* ASTERISK-27217 - chan_sip: Asterisk crashing when
subscription doesn't get set
(Reported by Bryan Walters)
* ASTERISK-17540 - SDP origin attribute modified when issuing
re-INVITE because of directmedia=yes
(Reported by saghul)
* ASTERISK-27254 - alembic: prune_on_boot fix erroneous
(Reported by Florian Floimair)
* ASTERISK-27232 - When in queue on g722 with interruptions,
music on hold can get stuck and no longer play
(Reported
by Jens T.)
* ASTERISK-27024 - nat/external_media settings ignored in
14.4.1
(Reported by Christopher van de Sande)
* ASTERISK-26879 - PJSIP external_media_address ignored if no
local_net options are provided
(Reported by Matt Jordan)
* ASTERISK-27236 - Segfault ast_channel_name (chan=0x0) at
channel_internal_api.c:478 during T.38 Fax Receive
(Reported by Ross Beer)
* ASTERISK-27225 - Crash when freeing dtls_cfg->cafile
(Reported by Richard Kenner)
* ASTERISK-27177 - ooh323c: misleading indentation in
addons/ooh323c/src/ooSocket.c
(Reported by Tzafrir Cohen)
* ASTERISK-27241 - libc segfault upon entry into app_directory
(Reported by David Moore)
* ASTERISK-27152 - Sending a "tel" uri in a From or To header
in an unauthenticated message causes asterisk to crash
(Reported by Ross Beer)
* ASTERISK-27103 - core: ast_safe_system command injection
possible.
(Reported by Corey Farrell)
* ASTERISK-27013 - res_rtp_asterisk: Media can be hijacked even
with strict RTP enabled
(Reported by Joshua Colp)
* ASTERISK-27231 - res_rtp_asterisk: Allow remote SSRC to
change due to renegotiation
(Reported by Joshua Colp)
* ASTERISK-26994 - Confbridge: CBAnn channels intermittently
become stuck when caller hangs up before recording name
(Reported by James Terhune)
* ASTERISK-27222 - core: Don't queue up multiple video update
frames.
(Reported by Joshua Colp)
* ASTERISK-20858 - app_minivm fails to clean up mkstemp files
(Reported by Walter Doekes)
* ASTERISK-16777 - several filename bugs in Record()
application
(Reported by klaus3000)
* ASTERISK-27168 - alembic: PJSIP scripts are missing column
dtls_fingerprint in ps_endpoints table
(Reported by
Florian Floimair)
* ASTERISK-27209 - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE
is used
(Reported by Torrey Searle)
* ASTERISK-19103 - When using realtime queues, function
QUEUE_MEMBER_LIST() will return an error if no other
app/function has loaded the queues first. This problem does not
exist if queues.conf is used.
(Reported by Jim Van
Meggelen)
* ASTERISK-21241 - When using voicemail as announce only
(maxmsg=0), the star dtmf to enter the voicemail is not honored
(Reported by Eelco Brolman)
* ASTERISK-27212 - bridge_softmix: Quickly joining/leaving may
cause video stream to remain in SFU
(Reported by Richard
Mudgett)
* ASTERISK-27204 - [patch] app_queue: Wrong queue stat
calculation
(Reported by sungtae kim)
* ASTERISK-27207 - XMPP OAuth not working due to inverted
logic
(Reported by Michael Kuron)
* ASTERISK-27174 - res_calendar_icalendar: Recurring events not
being loaded from Google calendar using ical
(Reported by
Mark Thompson)
* ASTERISK-27202 - If wget is not installed and "or" is not
available, external components (excluding pjsip) are not
installed
(Reported by Seán C. McCord)
* ASTERISK-27200 - manager: hook event is not being raised
(Reported by Kevin Harwell)
* ASTERISK-27147 - Either asterisk or pjproject isn't re-using
tcp connections (again)
(Reported by George Joseph)
* ASTERISK-27193 - IPv6 receive address in message doesn't
include brackets
(Reported by Scott Griepentrog)
* ASTERISK-27158 - [patch] res_rtp_asterisk: RTCP statistics
are not available when native bridge is used
(Reported by
Torrey Searle)
* ASTERISK-26745 - Asymmetric codecs when
asymmetric_rtp_codec=no
(Reported by Jesse Ross)
* ASTERISK-27189 - Make --with-pjproject-bundled the default
for Asterisk 15
(Reported by George Joseph)
* ASTERISK-27110 - RTP session is not fully destroyed on
channel hangup
(Reported by Matt Jordan)
* ASTERISK-27182 - bridge: Crash when mapping streams
(Reported by Joshua Colp)
* ASTERISK-27180 - channel: requester leaks joint_cap on
success.
(Reported by Corey Farrell)
* ASTERISK-27179 - res_pjsip_session: Handling of 'msid' is
incorrect
(Reported by Kevin Harwell)
* ASTERISK-27119 - res_pjsip: parse/add msid attribute when
webrtc is enabled
(Reported by Kevin Harwell)
* ASTERISK-27171 - Asterisk 15.0.0-Beta1 does not compile
(Reported by Ira Emus)
* ASTERISK-26659 - res_pjsip: PJSIP presence - missing braces
around the status element in XML
(Reported by Abraham
Liebsch)
* ASTERISK-27156 - Asterisk won't compile on Fedora 26 with
devmode enabled.
(Reported by Corey Farrell)
* ASTERISK-27001 - res_pjsip: TLS connection not stable
(Reported by Ian Gilmour)
* ASTERISK-27130 - Applications ARI: Unsubscribe action for
deviceStates does not remove old subscriptions properly
(Reported by Sergej Kasumovic)
* ASTERISK-25810 - say.c calls for sounds in the subdir
"digits" that don't exist (in Core). SayUnixTime or other Say...
apps will fail out when they call these sounds.
(Reported
by Nicolas Riendeau)
* ASTERISK-27142 - sounds: Conflict between files in
asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
(Reported by Corey Farrell)
* ASTERISK-27143 - bridge_softmix / res_rtp_asterisk: Fix
packet loss and renegotiation issues.
(Reported by Joshua
Colp)
Improvements made in this release:
-----------------------------------
* ASTERISK-22825 - Dialplan Function for Checking Parking Lot
Slot
(Reported by JoshE)
* ASTERISK-27912 - [PATCH] Add predial handler to app_queue
(Reported by Kristian HÞgh)
* ASTERISK-27929 - [patch] BuildSystem: Enable autotools in
Solaris 11.
(Reported by Alexander Traud)
* ASTERISK-27752 - Ten seconds of silence after mp3 playback
(Reported by Sam Wierema)
* ASTERISK-27910 - [patch] res_rtp_asterisk: Allow OpenSSL
configured with no-deprecated.
(Reported by Alexander
Traud)
* ASTERISK-27906 - [patch] res_crypto: Allow OpenSSL configured
with no-deprecated.
(Reported by Alexander Traud)
* ASTERISK-27877 - app_confbridge: Add talking indicator for
ConfBridgeList AMI response
(Reported by William McCall)
* ASTERISK-27873 - documentation: Error on wiki description of
Asterisk 13 "MeetmeMute" event
(Reported by Alessandro
Polidori)
* ASTERISK-27846 - ast_coredumper: Fix OUTPUT directory
(Reported by Ted G)
* ASTERISK-27867 - [patch] libasteriskssl: Allow OpenSSL 1.0.2
configured with no-deprecated.
(Reported by Alexander
Traud)
* ASTERISK-27796 - res_hep: Allow create_address to resolve a
provided hostname
(Reported by Sebastian Gutierrez)
* ASTERISK-27820 - [patch] Add DragonFly BSD.
(Reported
by Alexander Traud)
* ASTERISK-25129 - wrong automatic ras address assignment if
multihomed
(Reported by Dmitry Melekhov)
* ASTERISK-27793 - cppcheck identifies redundant "if"
(Reported by Ilya Shipitsin)
* ASTERISK-27697 - Enable in-dialog NOTIFY on chan_pjsip
channels
(Reported by Nathan Bruning)
* ASTERISK-27770 - [patch] install_prereq: Add Slackware
(somehow).
(Reported by Alexander Traud)
* ASTERISK-27769 - [patch] install_prereq: Add Gentoo Linux.
(Reported by Alexander Traud)
* ASTERISK-27738 - [patch] install_prereq: Add Arch Linux.
(Reported by Alexander Traud)
* ASTERISK-27736 - [patch] install_prereq: Add SUSE.
(Reported by Alexander Traud)
* ASTERISK-27253 - [patch] libsrtp-2.1.x support
(Reported by Alexander Traud)
* ASTERISK-27728 - [patch] BuildSystem: Add NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27730 - PJSIP: Update bundled PJPROJECT to version
2.7.2
(Reported by Richard Mudgett)
* ASTERISK-27729 - [patch] install_prereq: Add NetBSD.
(Reported by Alexander Traud)
* ASTERISK-27683 - [patch] BuildSystem: Allow newer autotools
on OpenBSD.
(Reported by Alexander Traud)
* ASTERISK-27348 - [patch]contrib/scripts: add a way to migrate
from chan_sip to chan_pjsip realtime
(Reported by Torrey
Searle)
* ASTERISK-27661 - Add new AMI Event for Load, Unload
(Reported by sungtae kim)
* ASTERISK-27651 - app_confbridge: Add Muted to ConfbridgeJoin
and channel snapshot headers to ConfbridgeList AMI events
(Reported by Richard Mudgett)
* ASTERISK-27647 - app_confbridge/bridge_softmix: When channel
muted report talking stopped if was talking.
(Reported by
Richard Mudgett)
* ASTERISK-27084 - Reduce verbosity while loading PBX
extensions.
(Reported by Ludovic Gasc (Eyepea))
* ASTERISK-24372 - [patch] Add config option to play a prompt
to the "winner" in app_followme
(Reported by Graham
Mainwaring)
* ASTERISK-27537 - res_pjsip: Add new AMI Action for
PJSIPShowAors
(Reported by sungtae kim)
* ASTERISK-27483 - Allow wrapuptime to be set for each queue
member
(Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-24297 - cdr.c: Minor code optimizations.
(Reported by Richard Mudgett)
* ASTERISK-27470 - Add new object for VoicemailUserEntry
(Reported by sungtae kim)
* ASTERISK-27461 - 3PCC patch for AMI "SIPnotify"
(Reported by Yasuhiko Kamata)
* ASTERISK-27449 - [PATCH] When failing to acquire target
during attended transfer, display wanted extension
(Reported by Niklas Larsson)
* ASTERISK-27456 - app_voicemail: Add new object for
VoicemailUserEntry
(Reported by sungtae kim)
* ASTERISK-27380 - ast_coredumper: allow pointing out the
asterisk binary explicitly
(Reported by Tzafrir Cohen)
* ASTERISK-23556 - Compilation warning for invert.c (array
subscript is above array bounds)
(Reported by Marcello
Ceschia)
* ASTERISK-27359 - pjproject bundled: Don't disable assertions
when --enable-dev-mode is used.
(Reported by Corey
Farrell)
* ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
(Reported by Richard Mudgett)
* ASTERISK-27335 - CDR performance needs improvement.
(Reported by Richard Mudgett)
* ASTERISK-27278 - [patch] chan_sip: Provide access to read the
full SIP Request-URI from INVITE
(Reported by David J.
Pryke)
* ASTERISK-27255 - alembic: Add support for Microsoft SQL
server
(Reported by Florian Floimair)
* ASTERISK-27220 - Enable CHANNEL function to get from and to
tag from SIP Headers
(Reported by Andre Nazario)
* ASTERISK-27169 - Google OAuth 2.0 support for XMPP / Motif
(Reported by Andrey)
* ASTERISK-27173 - Support for GMIME 3.0
(Reported by
Tzafrir Cohen)
* ASTERISK-27085 - [patch] chan_pjsip: Port SIPDtmfMode to
chan_pjsip
(Reported by Torrey Searle)
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0-rc1
Thank you for your continued support of Asterisk!