[asterisk-dev] Asterisk 16.0.0 Now Available
Asterisk Development Team
asteriskteam at digium.com
Tue Oct 9 12:40:36 CDT 2018
The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.0.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Security bugs fixed in this release:
-----------------------------------
* ASTERISK-28013 - res_http_websocket: Crash when reading HTTP
Upgrade requests
(Reported by Sean Bright)
* 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-28033 - AMI event "NewExten" is set to the wrong
class
(Reported by lvl)
* ASTERISK-27988 - alembic: PJSIP
"mwi_subscribe_replaces_unsolicited" field is integer not
boolean
(Reported by Joshua C. Colp)
* ASTERISK-28022 - res_pjsip realtime: uri column in
ps_contacts table can be too short
(Reported by Florian
Floimair)
* ASTERISK-27978 - res_pjsip: Change default transport
keepalive to preserve behavior
(Reported by Joshua C.
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 C. Colp)
* ASTERISK-27965 - module: Remove old modules, update support
levels
(Reported by Joshua C. Colp)
* ASTERISK-25548 - stasis: Improve message type "Use of before
init/after destruction" error
(Reported by Joshua C.
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-27944 - res_pjsip_t38: Crash receiving 1xx responses
other than 100 before 200 for T.38 reINVITE
(Reported by
Joshua Elson)
* 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 C. 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 C. 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 C��sar 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 C. 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 C. 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 C. Colp)
* ASTERISK-26366 - rtp: RTCP messages with REMB trigger fast
picture update
(Reported by Joshua C. 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 S��bastien 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 Nem��i��)
* 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 S��bastien 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 C.
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 St��phan 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 C.
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 Pr��)
* ASTERISK-25128 - Datastore: Implement automatic module
references.
(Reported by Corey Farrell)
* ASTERISK-27366 - Asterisk Turkish Language Set Problem
(Reported by Halil ��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 C. 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 C. 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 C. 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 Cis��rik)
* 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 C.
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 Beno��t Dereck-Tricot)
* ASTERISK-27216 - app_queue: does its
check-makeannouncement-logic twice each head-caller-loop
(Reported by Stefan Engstr��m)
* 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 C. 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 C. Colp)
* ASTERISK-27231 - res_rtp_asterisk: Allow remote SSRC to
change due to renegotiation
(Reported by Joshua C. 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 C. 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 Se��n 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 C. 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
C. Colp)
Improvements made in this release:
-----------------------------------
* ASTERISK-27164 - [patch] Add IPv6 Support for DUNDi
(Reported by Adam Secombe)
* ASTERISK-22825 - Dialplan Function for Checking Parking Lot
Slot
(Reported by JoshE)
* ASTERISK-27912 - [PATCH] Add predial handler to app_queue
(Reported by Kristian H��gh)
* 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, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
Thank you for your continued support of Asterisk!
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