VoIP Mailing List Archives

[asteriskteam at digium...]

The Asterisk Development Team has announced the first beta of
Asterisk 14.0.0. This beta is available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 14.0.0-beta1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this beta:

New Features made in this release:
-----------------------------------
* ASTERISK-25904 - PJSIP: add contact.updated event (Reported by
Alexei Gradinari)
* ASTERISK-26058 - [Patch] Add uptime and last reloaded to
FullyBooted AMI event (Reported by Niklas Larsson)
* ASTERISK-25925 - Allow Early Bridges on ARI Dials (Reported by
Mark Michelson)
* ASTERISK-26068 - Multicast RTP Options (Reported by Mark
Michelson)
* ASTERISK-26042 - ARI: Allow downloading of the media associated
with a stored recording (Reported by Matt Jordan)
* ASTERISK-25425 - logger: Add JSON structured logging (Reported
by Matt Jordan)
* ASTERISK-25900 - PJSIP Endpoint IP Access Controls (Reported by
Alexei Gradinari)
* ASTERISK-25972 - res_pjsip_exten_state: Use body generator to
publish extension state (Reported by Richard Mudgett)
* ASTERISK-25889 - ARI: Add separate "create" and "dial"
operations for channels (Reported by Mark Michelson)
* ASTERISK-25803 - [patch] chan_sip: Optionally supply
fromuser/fromdomain in SIP dial string (Reported by Walter
Doekes)
* ASTERISK-24919 - res_pjsip_config_wizard: Ability to write
contents to file (Reported by Ray Crumrine)
* ASTERISK-25670 - Add regcontext to PJSIP (Reported by Daniel
Journo)
* ASTERISK-25660 - Add sipp-sendfax.xml and spandspflow2pcap.py to
contrib/scripts. (Reported by Walter Doekes)
* ASTERISK-25591 - [patch] Complete List of Header Files
(#include): iwyu (Reported by Alexander Traud)
* ASTERISK-25551 - [patch]Ability to add channel to an existing
bridge by specifying an existing channel prefix (Reported by
Alec Davis)
* ASTERISK-25419 - Dialplan Application for Integration of StatsD
(Reported by Ashley Sanders)
* ASTERISK-25549 - Confbridge: Add participant timeout option
(Reported by Mark Michelson)
* ASTERISK-24922 - ARI: Add the ability to intercept hold and
raise an event (Reported by Matt Jordan)
* ASTERISK-25479 - Allow CDR's to be modified before being
dispatched to engines (Reported by Jonh Wendell)
* ASTERISK-25480 - [patch]Add field PauseReason on
QueueMemberStatus (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25377 - res_pjsip: Change default "From user" from UUID
to something more palatable (Reported by Mark Michelson)
* ASTERISK-25252 - ARI: Add the ability to manipulate log channels
(Reported by Matt Jordan)
* ASTERISK-25259 - chan_pjsip: Add rtptimeout support (Reported by
Joshua Colp)
* ASTERISK-25238 - ARI: Support push configuration (Reported by
Matt Jordan)
* ASTERISK-25173 - ARI: Add the ability to load/reload/unload an
Asterisk module (Reported by Matt Jordan)
* ASTERISK-25006 - [patch] Add support set character for quoted
identifiers (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-23186 - [patch] Add usegmtime option to cel_pgsql
(Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-24931 - dns: Add support for SRV records. (Reported by
Joshua Colp)
* ASTERISK-24834 - DNS Overhaul: Implement the proposed core API -
sync/async functions, resolver registration (Reported by Matt
Jordan)
* ASTERISK-24836 - DNS Overhaul: Write a Resolver Implementation
(Reported by Matt Jordan)
* ASTERISK-22591 - [patch]Prevent Asterisk from writing received
SMS content in log (Reported by Jan Juergens)
* ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
(Reported by Dwayne Hubbard)
* ASTERISK-24703 - ARI: Add the ability to "transfer" (redirect) a
channel (Reported by Matt Jordan)
* ASTERISK-24363 - [patch] Add ability for Channel Drivers to
provide Presence State information (Reported by Gareth Palmer)
* ASTERISK-24554 - AMI/ARI: Generate events on connected line
changes (Reported by Matt Jordan)
* ASTERISK-24276 - [Patch] Option to make app MOH override channel
musicclass (Reported by Kristian Høgh)
* ASTERISK-23871 - RLS Tests: Implement RLS off-nominal tests
(Reported by Mark Michelson)
* ASTERISK-23823 - [patch] Option to keep queuerules in realtime
(Reported by Michael K.)

Bugs fixed in this release:
-----------------------------------
* ASTERISK-26227 - sqlalchemy error due to long identifier name
(Reported by Mark Michelson)
* ASTERISK-26221 - chan_sip: iLBC does not include correct mode
(Reported by Aaron Meriwether)
* ASTERISK-26216 - res_fax: Deadlock when detect fax while channel
executing Playback (Reported by Richard Mudgett)
* ASTERISK-26214 - Allow arbitrary time for fax detection to end
on a channel (Reported by Richard Mudgett)
* ASTERISK-23013 - [patch] Deadlock between 'sip show channels'
command and attended transfer handling (Reported by Ben
Smithurst)
* ASTERISK-26212 - [patch] Makefile: Retain XML Declaration and
DTD in docs. (Reported by Alexander Traud)
* ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in
conditional code. (Reported by Corey Farrell)
* ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence
number even on lost packets. (Reported by Alexander Traud)
* ASTERISK-26038 - 'make install' doesn't seem to install OS/X
init files (Reported by Tzafrir Cohen)
* ASTERISK-26133 - app_queue: Queue members receive multiple calls
(Reported by Richard Miller)
* ASTERISK-26196 - pbx: Time based includes can leak timezone
string (Reported by Corey Farrell)
* ASTERISK-26193 - chan_sip: reference leak in mwi_event_cb
(Reported by Corey Farrell)
* ASTERISK-26191 - threadpool: Leak on duplicate taskprocessor for
ast_threadpool_serializer_group (Reported by Corey Farrell)
* ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing
DTLS failure occurred on RTP instance (Reported by Edwin
Vandamme)
* ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf.
(Reported by Alexander Traud)
* ASTERISK-26160 - pjsip: Updated->Reachable during qualify
(Reported by Matt Jordan)
* ASTERISK-26177 - func_odbc: Database handle is kept when it
should be released (Reported by Leandro Dardini)
* ASTERISK-25289 - Build System does not respect CFLAGS and
CXXFLAGS when building menuselect (Reported by Jeffrey Walton)
* ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out
of bounds and bugs (Reported by Alexei Gradinari)
* ASTERISK-26184 - chan_sip: Reference leaks in error paths.
(Reported by Corey Farrell)
* ASTERISK-26181 - REF_DEBUG: Node object incorrectly logged
during duplicate replacement (Reported by Corey Farrell)
* ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported
by Joshua Colp)
* ASTERISK-26180 - PJSIP: provide valid tcp nodelay option for
reuse (Reported by Scott Griepentrog)
* ASTERISK-25772 - res_pjsip: Unexpected two BYE when answered
(Reported by Dmitriy Serov)
* ASTERISK-26099 - res_pjsip_pubsub: Crash when sending request
due to server timeout (Reported by Ross Beer)
* ASTERISK-26144 - Crash on loading codecs g729/g723 (Reported by
Alexei Gradinari)
* ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x
(Reported by George Joseph)
* ASTERISK-26021 - Build codecs siren7 and siren14 for Asterisk 13
(Reported by Daniel Denson)
* ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to
v21_details (Reported by Corey Farrell)
* ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a
self-comparison (Reported by George Joseph)
* ASTERISK-26138 - chan_unistim: Under FreeBSD, chan_unistim
generates a compile error (Reported by George Joseph)
* ASTERISK-26128 - Alembic scripts are failing (Reported by Mark
Michelson)
* ASTERISK-26139 - test_res_pjsip_scheduler: Compile failure if
pjproject isn't installed in a system location (Reported by
George Joseph)
* ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version.
(Reported by Alexander Traud)
* ASTERISK-26132 - PJSIP: provide transport type with received
messages (Reported by Scott Griepentrog)
* ASTERISK-26127 - res_pjsip_session: Crash due to race condition
between res_pjsip_session unload and timer (Reported by Joshua
Colp)
* ASTERISK-26045 - [patch]app_voicemail: fix bugs, imap mm_status
log change to debug (Reported by Alexei Gradinari)
* ASTERISK-26083 - ARI: Announcer channels staying around after
playback to a bridge is finished (Reported by Per Jensen)
* ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in
http.conf (Reported by Alexander Traud)
* ASTERISK-26097 - [patch] CLI: show maximum file descriptors
(Reported by Alexander Traud)
* ASTERISK-25262 - Memory leak when a caller channel does multiple
dials and CEL is enabled (Reported by Etienne Lessard)
* ASTERISK-26047 - ARI allows certain commands to run on down
channels. (Reported by Mark Michelson)
* ASTERISK-25959 -
http_media_cache/retrieve_cache_control_directives: Sporadic
failure (Reported by Joshua Colp)
* ASTERISK-26103 - cdr: Assert on 'dial end' event during a blond
transfer (Reported by George Joseph)
* ASTERISK-26092 - [Segfault] in res_rtp_asterisk.c:4268 after
Remotely bridged channels (Reported by Niklas Larsson)
* ASTERISK-26089 - Invalid security events during boot using PJSIP
Realtime (Reported by Scott Griepentrog)
* ASTERISK-26096 - res_hep: Crash when configuration file is
missing (Reported by Niklas Larsson)
* ASTERISK-26074 - res_odbc: Deadlock within UnixODBC (Reported by
Ross Beer)
* ASTERISK-26054 - Asterisk crashes (core dump) (Reported by B.
Davis)
* ASTERISK-26069 - Asterisk truncates To: header, dropping the
closing '>' (Reported by Vasil Kolev)
* ASTERISK-24436 - Missing header in res/res_srtp.c when compiling
against libsrtp-1.5.0 (Reported by Patrick Laimbock)
* ASTERISK-26091 - [patch] ar cru creates warning, instead use ar
cr (Reported by Alexander Traud)
* ASTERISK-26070 - ari/channels: Creating a local channel without
an originator adds all audio formats to it's capabilities
(Reported by George Joseph)
* ASTERISK-26078 - core: Memory leak in logging (Reported by
Etienne Lessard)
* ASTERISK-26065 - chan_pjsip: MWI NOTIFY contents not ordered
properly (Reported by Ross Beer)
* ASTERISK-26063 - ${PJSIP_HEADER(read,Call-ID)} does not work -
documentation needs clarification for when read/write is
possible (Reported by Private Name)
* ASTERISK-25777 - data race in threadpool (Reported by Badalian
Vyacheslav)
* ASTERISK-26053 - res_pjsip_outbound_publish: Crash when shutting
down (Reported by Joshua Colp)
* ASTERISK-26049 - res_pjsip: Crash when our own request timer
fires (Reported by Joshua Colp)
* ASTERISK-25669 - [patch]CURL incorrect trim for non ASCII
characters (Reported by Jesper)
* ASTERISK-26029 - parking: ast_parking_park_call should return
parking_space instead of parking_exten (Reported by Diederik de
Groot)
* ASTERISK-25938 - res_odbc: MySQL/MariaDB statement
LAST_INSERT_ID() always returns zero. (Reported by Edwin
Vandamme)
* ASTERISK-25941 - chan_pjsip: Crash on an immediate SIP final
response (Reported by Javier Riveros )
* ASTERISK-26014 - res_sorcery_astdb: Make tolerant of unknown
fields (Reported by Joshua Colp)
* ASTERISK-24986 - keepalive INFO packages ignored by asterisk
(Reported by Ilya Trikoz)
* ASTERISK-26034 - T.38 passthrough problem behind firewall due to
early nosignal packet (Reported by George Joseph)
* ASTERISK-26030 - call cut because of double Session-Expires
header in re-invite after proxy authentication is required
(Reported by George Joseph)
* ASTERISK-25964 - Outbound registrations created via ARI/push
configuration do not clean up outbound registrations currently
in flight (Reported by Matt Jordan)
* ASTERISK-26005 - res_pjsip: Multiple SIP messages are combined
into 1 TCP packet (Reported by Ross Beer)
* ASTERISK-25352 - res_hep_rtcp correlation_id is different then
res_hep (Reported by Kevin Scott Adams)
* ASTERISK-26007 - res_pjsip: Endpoints deleting early after
upgrade from 13.8.2 to 13.9 (Reported by Greg Siemon)
* ASTERISK-25990 - PJSIP TLS registration should respect
client_uri scheme when generating Contact URI (Reported by
Sebastian Damm)
* ASTERISK-26008 - app_followme does not delete recorded name
prompt (Reported by Tzafrir Cohen)
* ASTERISK-25978 - res_pjsip_authenticator_digest: Should not use
source port in nonce verification (Reported by Mark Michelson)
* ASTERISK-26004 - res_pjsip: The transport/method parameter is
ignored (Reported by George Joseph)
* ASTERISK-25999 - res_pjsip_dialog_info_body_generator: Remove
subscription requirement (Reported by Joshua Colp)
* ASTERISK-25993 - pjproject: Allow bundling to not require
everything it does (Reported by Joshua Colp)
* ASTERISK-25998 - file: Crash when using nativeformats (Reported
by Joshua Colp)
* ASTERISK-25826 - PJSIP / Sorcery slow load from realtime
(Reported by Ross Beer)
* ASTERISK-25956 - Compilation error in conditionally compiled
code in config_options.c (Reported by Chris Trobridge)
* ASTERISK-25968 - pjproject_bundled: Configure and make need to
be re-tested (Reported by George Joseph)
* ASTERISK-24463 - Voicemail email address corrupt or not sent
when message is in the process of being recorded during reload
(Reported by John Campbell)
* ASTERISK-25922 - res_pjsip_exten_state: Add configuration
support for publishing (Reported by Joshua Colp)
* ASTERISK-25970 - Segfault in pjsip_url_compare (Reported by
Dmitriy Serov)
* ASTERISK-25963 - func_odbc requires reconnect checks for stale
connections (Reported by Ross Beer)
* ASTERISK-25961 - tests/channels/SIP/sip_tls_call: Sporadic crash
when running test (Reported by Joshua Colp)
* ASTERISK-16115 - [patch] problem with ringinuse=no, queue
members receive sometimes two calls (Reported by nik600)
* ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir
only works if you manually add secret.conf yourself (Reported by
Jonathan R. Rose)
* ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions
are case sensitive to QueueName (Reported by Javier Acosta)
* ASTERISK-25951 - res_agi: run_agi eats frames it shouldn't
(Reported by George Joseph)
* ASTERISK-25950 - [patch]SIP channel does not send PeerStatus
events for autocreated peers (Reported by Kirill Katsnelson)
* ASTERISK-25927 - Removed option "registertrying" is still
documented in sip.conf.sample (Reported by Etienne Lessard)
* ASTERISK-25947 - Protocol transfers to stasis applications are
missing the StasisStart with the replace_channel object.
(Reported by Richard Mudgett)
* ASTERISK-24649 - Pushing of channel into bridge fails; Stasis
fails to get app name (Reported by John Bigelow)
* ASTERISK-24782 - StasisEnd event not present for channel that
was swapped out for another after completing attended transfer
(Reported by John Bigelow)
* ASTERISK-25942 - res_pjsip_caller_id: Transfer results in mixed
ConnectedLine information (Reported by George Joseph)
* ASTERISK-25928 - res_pjsip: URI validation done outside of PJSIP
thread (Reported by Joshua Colp)
* ASTERISK-25929 - res_pjsip_registrar: AOR_CONTACT_ADDED events
not raised (Reported by Joshua Colp)
* ASTERISK-25934 - chan_sip should not require sipregs or
updateable sippeers table unless rt (Reported by Jaco Kroon)
* ASTERISK-25888 - Frequent segfaults in function can_ring_entry()
of app_queue.c (Reported by Sébastien Couture)
* ASTERISK-25796 - res_pjsip: DOS/Crash when TCP/TLS sockets
exceed pjproject PJ_IOQUEUE_MAX_HANDLES (Reported by George
Joseph)
* ASTERISK-25707 - Long contact URIs or hostnames can crash
pjproject/Asterisk under certain conditions (Reported by George
Joseph)
* ASTERISK-25123 - Bracketed IPv6 Contact header parameter
unparsable with Asterisk/PJSIP (Reported by Anthony Messina)
* ASTERISK-25874 - app_voicemail: Stack buffer overflow in
test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
* ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE
without adding them to the local hangupcauses via
ast_channel_hangupcause_hash_set (Reported by Jaco Kroon)
* ASTERISK-25885 - res_pjsip: Race condition between adding
contact and automatic expiration (Reported by Joshua Colp)
* ASTERISK-25910 - pjproject: Via headers are not parsed when
"received" contains an IPv6 address (Reported by George Joseph)
* ASTERISK-25890 - Asterisk 13.8.0 alembic database update fails
(Reported by Harley Peters)
* ASTERISK-25894 - [patch] webrtc video broken due to missing
marker bits in RTP streams (Reported by Jacek Konieczny)
* ASTERISK-25881 - pbx: Add support for autohints (Reported by
Joshua Colp)
* ASTERISK-25854 - No audio after HOLD/RESUME - incorrect
a=recvonly in SDP from Asterisk (Reported by Robert McGilvray)
* ASTERISK-25868 - Sorcery "append to category" should allow
filters (Reported by Nick Repin)
* ASTERISK-25873 - res_pjsip: Bundled pjproject: compile error,
cannot find -lasteriskpj (Reported by Hans van Eijsden)
* ASTERISK-25882 - ARI: Crash can occur due to race condition when
attempting to operate on a hung up channel (Part 2) (Reported by
Richard Mudgett)
* ASTERISK-25867 - [patch] Video delay on app_echo (Reported by
Jacek Konieczny)
* ASTERISK-24605 - res_parking option parkeddynamic does not work
with the core Features 'parkcall' (DTMF initiated parking)
(Reported by Philip Correia)
* ASTERISK-24596 - Unclear how to use Park application with
res_parking 'parkeddynamic' enabled. Documentation? (Reported by
Philip Correia)
* ASTERISK-25825 - Crashes during shutdown when running CLI
commands (Reported by Mark Michelson)
* ASTERISK-24543 - Asterisk 13 responds to SIP Invite with all
possible codecs configured for peer as opposed to intersection
of configured codecs and offered codecs (Reported by Taylor
Hawkes)
* ASTERISK-25407 - Asterisk fails to log to multiple syslog
destinations (Reported by Elazar Broad)
* ASTERISK-25510 - [patch]Log to syslog failing (Reported by
Michael Newton)
* ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
data corruption (Reported by Gianluca Merlo)
* ASTERISK-25849 - chan_pjsip: transfers with direct media
sometimes drops audio (Reported by Kevin Harwell)
* ASTERISK-25814 - Segfault at f ip in res_pjsip_refer.so
(Reported by Sergio Medina Toledo)
* ASTERISK-25023 - Deadlock in chan_sip in
update_provisional_keepalive (Reported by Arnd Schmitter)
* ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
channel (Reported by Filip Frank)
* ASTERISK-25829 - res_pjsip: PJSIP does not accept spaces when
separating multiple AORs (Reported by Mateusz Kowalski)
* ASTERISK-25771 - ARI:Crash - Attended transfers of channels into
Stasis application. (Reported by Javier Riveros )
* ASTERISK-25830 - Revision 2451d4e breaks NAT (Reported by Sean
Bright)
* ASTERISK-25582 - Testsuite: Reactor timeout error in
tests/fax/pjsip/directmedia_reinvite_t38 (Reported by Matt
Jordan)
* ASTERISK-25811 - Unable to delete object from sorcery cache
(Reported by Ross Beer)
* ASTERISK-25800 - [patch] Calculate talktime when is first call
answered (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25727 - RPM build requires OPTIONAL_API cflag due to
PJSIP requirement (Reported by Gergely Dömsödi)
* ASTERISK-25337 - Crash on PJSIP_HEADER Add P-Asserted-Identity
when calling from Gosub (Reported by Jacques Peacock)
* ASTERISK-25738 - res_pjsip_pubsub: Crash while executing
OutboundSubscriptionDetail ami action (Reported by Kevin
Harwell)
* ASTERISK-25721 - [patch] res_phoneprov: memory leak and
heap-use-after-free (Reported by Badalian Vyacheslav)
* ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
returns garbage (Reported by Etienne Lessard)
* ASTERISK-25751 - res_pjsip: Support
pjsip_dlg_create_uas_and_inc_lock (Reported by Joshua Colp)
* ASTERISK-25606 - Core dump when using transports in sorcery
(Reported by Martin Moučka)
* ASTERISK-20987 - non-admin users, who join muted conference are
not being muted (Reported by hristo)
* ASTERISK-25737 - res_pjsip_outbound_registration: line option
not in Alembic (Reported by Joshua Colp)
* ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
Vulnerability - Investigate vulnerability of HTTP server
(Reported by Alex A. Welzl)
* ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
udptl_rx_packet cause ast_frdup crash (Reported by Walter
Doekes)
* ASTERISK-25742 - Secondary IFP Packets can result in accessing
uninitialized pointers and a crash (Reported by Torrey Searle)
* ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
non-default timert1 (Reported by Alexander Traud)
* ASTERISK-25702 - PjSip realtime DB and Cache Errors since
upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported by
Nic Colledge)
* ASTERISK-25730 - build: make uninstall after make distclean
tries to remove root (Reported by George Joseph)
* ASTERISK-25725 - core: Incorrect XML documentation may result in
weird behavior (Reported by Joshua Colp)
* ASTERISK-25722 - ASAN & testsute: stack-buffer-overflow in
sip_sipredirect (Reported by Badalian Vyacheslav)
* ASTERISK-25709 - ARI: Crash can occur due to race condition when
attempting to operate on a hung up channel (Reported by Mark
Michelson)
* ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
by Badalian Vyacheslav)
* ASTERISK-25685 - infrastructure: Run alembic in Jenkins build
script (Reported by Joshua Colp)
* ASTERISK-25712 - Second call to already-on-call phone and
Asterisk sends "Ready" (Reported by Richard Mudgett)
* ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
(Reported by Badalian Vyacheslav)
* ASTERISK-25179 - CDR(billsec,f) and CDR(duration,f) report
incorrect values (Reported by Gianluca Merlo)
* ASTERISK-25611 - core: threadpool thread_timeout_thrash unit
test sporadically failing (Reported by Joshua Colp)
* ASTERISK-25686 - PJSIP: qualify_timeout is a double, database
schema is an integer (Reported by Marcelo Terres)
* ASTERISK-25700 - main/config: Clean config maps on shutdown.
(Reported by Corey Farrell)
* ASTERISK-25696 - bridge_basic: don't cache xferfailsound during
a transfer (Reported by Kevin Harwell)
* ASTERISK-25697 - bridge_basic: don't play an attended transfer
fail sound after target hangs up (Reported by Kevin Harwell)
* ASTERISK-25683 - res_ari: Asterisk fails to start if compiled
with MALLOC_DEBUG (Reported by yaron nahum)
* ASTERISK-24097 - Documentation - CHANNEL function help text
missing 'linkedid' argument (Reported by Steven T. Wheeler)
* ASTERISK-25690 - Hanging up when executing connected line sub
does not cause hangup (Reported by Joshua Colp)
* ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
reload' cause a crash (Reported by Sean Bright)
* ASTERISK-25632 - res_pjsip_sdp_rtp: RTP is sent from wrong IP
address when multihomed (Reported by Olivier Krief)
* ASTERISK-25637 - Multi homed server using wrong IP (Reported by
Daniel Journo)
* ASTERISK-25394 - pbx: Incorrect device and presence state when
changing hint details (Reported by Joshua Colp)
* ASTERISK-25640 - pbx: Deadlock on features reload and state
change hint. (Reported by Krzysztof Trempala)
* ASTERISK-25681 - devicestate: Engine thread is not shut down
(Reported by Corey Farrell)
* ASTERISK-25680 - manager: manager_channelvars is not cleaned at
shutdown (Reported by Corey Farrell)
* ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
Corey Farrell)
* ASTERISK-25675 - Endpoint not listed as Unreachable (Reported by
Daniel Journo)
* ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
by Corey Farrell)
* ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
Farrell)
* ASTERISK-25668 - res_pjsip: Deadlock in distributor (Reported by
Mark Michelson)
* ASTERISK-25664 - ast_format_cap_append_by_type leaks a reference
(Reported by Corey Farrell)
* ASTERISK-25647 - bug of cel_radius.c: wrong point of
ADD_VENDOR_CODE (Reported by Aaron An)
* ASTERISK-25137 - endpoint stasis messages are delivered twice
(Reported by Vitezslav Novy)
* ASTERISK-25116 - res_pjsip: Two PeerStatus AMI messages are
sent for every status change (Reported by George Joseph)
* ASTERISK-25641 - bridge: GOTO_ON_BLINDXFR doesn't work on
transfer initiated channel (Reported by Dmitry Melekhov)
* ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
Brandon)
* ASTERISK-25625 - res_sorcery_memory_cache: Add full backend
caching (Reported by Joshua Colp)
* ASTERISK-25601 - json: Audit reference usage and thread safety
(Reported by Joshua Colp)
* ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
sungtae kim)
* ASTERISK-25615 - res_pjsip: Setting transport async_operations >
1 causes segfault on tls transports (Reported by George Joseph)
* ASTERISK-25442 - using realtime (mysql) queue members are never
updated in wait_our_turn function (app_queue.c) (Reported by
Carlos Oliva)
* ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
thread of asterisk is not released (Reported by Hiroaki Komatsu)
* ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
Corey Farrell)
* ASTERISK-25619 - res_chan_stats not sending the correct
information to StatsD (Reported by Tyler Cambron)
* ASTERISK-24146 - [patch]No audio on WebRtc caller side when
answer waiting time is more than ~7sec (Reported by Aleksei
Kulakov)
* ASTERISK-25609 - [patch]Asterisk may crash when calling
ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
* ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
(Reported by Alexander Traud)
* ASTERISK-25616 - Warning with a Codec Module which supports PLC
with FEC (Reported by Alexander Traud)
* ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
Dudás József)
* ASTERISK-25608 - res_pjsip/contacts/statsd: Lifecycle events
aren't consistent (Reported by George Joseph)
* ASTERISK-25584 - [patch] format-attribute module: VP8 missing
(Reported by Alexander Traud)
* ASTERISK-25583 - [patch] format-attribute module: RFC 7587 (Opus
Codec) (Reported by Alexander Traud)
* ASTERISK-25498 - Asterisk crashes when negotiating g729 without
that module installed (Reported by Ben Langfeld)
* ASTERISK-25595 - Unescaped : in messge sent to statsd (Reported
by Niklas Larsson)
* ASTERISK-25598 - res_pjsip: Contact status messages are
printing a hash instead of the uri (Reported by George Joseph)
* ASTERISK-25600 - bridging: Inconsistency in BRIDGEPEER (Reported
by Jonathan Rose)
* ASTERISK-25476 - chan_sip loses registrations after a while
(Reported by Michael Keuter)
* ASTERISK-25593 - fastagi: record file closed after sending
result (Reported by Kevin Harwell)
* ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
it's assumed to (Reported by Walter Doekes)
* ASTERISK-25590 - CLI Usage info for 'pjsip send notify'
references incorrect config (Reported by Corey Farrell)
* ASTERISK-25165 - Testsuite - Sorcery memory cache leaks
(Reported by Corey Farrell)
* ASTERISK-25575 - res_pjsip: Dynamic outbound registrations
created via ARI are not loaded into memory on Asterisk
start/restart (Reported by Matt Jordan)
* ASTERISK-25545 - [patch] translation module gets cached not
joint format (Reported by Alexander Traud)
* ASTERISK-25573 - [patch] H.264 format attribute module: resets
whole SDP (Reported by Alexander Traud)
* ASTERISK-24958 - Forwarding loop detection inhibits certain
desirable scenarios (Reported by Mark Michelson)
* ASTERISK-25561 - app_queue.c line 6503 (try_calling): mutex
'qe->chan' freed more times than we've locked! (Reported by Alec
Davis)
* ASTERISK-25565 - DNS: System resolver only returns 1 record per
result (Reported by George Joseph)
* ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
Joshua Colp)
* ASTERISK-25160 - [patch] Opus Codec: SIP/SDP line fmtp missing
when called internally (Reported by Alexander Traud)
* ASTERISK-25535 - [patch] format creation on module load instead
of cache (Reported by Alexander Traud)
* ASTERISK-25449 - main/sched: Regression introduced by
5c713fdf18f causes erroneous duplicate RTCP messages; other
potential scheduling issues in chan_sip/chan_skinny (Reported by
Matt Jordan)
* ASTERISK-25546 - threadpool: Race condition between idle timeout
and activation (Reported by Joshua Colp)
* ASTERISK-25537 - [patch] format-attribute module: RFC or
internal defaults? (Reported by Alexander Traud)
* ASTERISK-25533 - [patch] buffer for ast_format_cap_get_names
only 64 bytes (Reported by Alexander Traud)
* ASTERISK-25373 - add documentation for CALLERID(pres) and also
the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
Doekes)
* ASTERISK-25528 - DNS: System resolver issues with TTL parse
(Reported by dtryba)
* ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
Walter Doekes)
* ASTERISK-24779 - Passthrough OPUS codec not working with
chan_pjsip (Reported by PowerPBX)
* ASTERISK-25522 - ARI: Crash when creating channel via ARI
originate with requesting channel (Reported by Matt Jordan)
* ASTERISK-25434 - Compiler flags not reported in 'core show
settings' despite usage during compilation (Reported by Rusty
Newton)
* ASTERISK-24106 - WebSockets Automatically decides what driver it
will use (Reported by Andrew Nagy)
* ASTERISK-25513 - Crash: malloc failed with high load of
subscriptions. (Reported by John Bigelow)
* ASTERISK-25505 - res_pjsip_pubsub: Crash on off-nominal when UAS
dialog can't be created (Reported by Joshua Colp)
* ASTERISK-25494 - build: GCC 5.1.x catches some new const, array
bounds and missing paren issues (Reported by George Joseph)
* ASTERISK-25485 - res_pjsip_outbound_registration: registration
stops due to 400 response (Reported by Kevin Harwell)
* ASTERISK-25486 - res_pjsip: Fix deadlock when validating URIs
(Reported by Joshua Colp)
* ASTERISK-7803 - [patch] Update the maximum packetization values
in frame.c (Reported by dea)
* ASTERISK-25484 - [patch] autoframing=yes has no effect (Reported
by Alexander Traud)
* ASTERISK-25308 - ari: Websocket leak (Reported by Joshua Colp)
* ASTERISK-25461 - Nested dialplan #includes don't work as
expected. (Reported by Richard Mudgett)
* ASTERISK-25455 - Deadlock of PJSIP realtime over
res_config_pgsql (Reported by mdu113)
* ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
(Reported by Olle Johansson)
* ASTERISK-25108 - configure check for older unbound library
(Reported by John Bigelow)
* ASTERISK-25435 - Asterisk periodically hangs. UDP Recv-Q greatly
exceeds zero. (Reported by Dmitriy Serov)
* ASTERISK-25451 - Broken video - erased rtp marker bit (Reported
by Stefan Engström)
* ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
exist in AstDB (Reported by Andrew Nagy)
* ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
parsing (Reported by ffs)
* ASTERISK-25404 - segfault/crash in chan_pjsip_hangup ... at
chan_pjsip.c (Reported by Chet Stevens)
* ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
(Reported by Bojan Nemčić)
* ASTERISK-25441 - Deadlock in res_sorcery_memory_cache. (Reported
by Richard Mudgett)
* ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
ICE is not enabled (Reported by Joshua Colp)
* ASTERISK-25383 - Core dumps on startup and shutdown with
MALLOC_DEBUG enabled (Reported by yaron nahum)
* ASTERISK-25423 - Caller gets no Connected line update during
call pickup. (Reported by Richard Mudgett)
* ASTERISK-25305 - Dynamic logger channels can be added multiple
times (Reported by Mark Michelson)
* ASTERISK-25418 - On-hold channels redirected out of a bridge
appear to still be on hold (Reported by Mark Michelson)
* ASTERISK-25384 - Regular Asterisk crashes when using Page
application. "user_data is NULL" (Reported by Chet Stevens)
* ASTERISK-25410 - app_record: RECORDED_FILE variable not being
populated (Reported by Kevin Harwell)
* ASTERISK-25396 - chan_sip: Extremely long callerid name causes
invalid SIP (Reported by Walter Doekes)
* ASTERISK-25399 - app_queue: AgentComplete event has wrong reason
(Reported by Kevin Harwell)
* ASTERISK-25185 - Segfault in app_queue on transfer scenarios
(Reported by Etienne Lessard)
* ASTERISK-25353 - [patch] Transcoding while different in Frame
size = Frames lost (Reported by Alexander Traud)
* ASTERISK-25325 - ARI PUT reload chan_sip HTTP response 404
(Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25390 - default_from_user can crash with certain
configuration backends (Reported by Mark Michelson)
* ASTERISK-25387 - res_pjsip_nat: Malformed REGISTER request
causes NAT'd Contact header to not be rewritten (Reported by
Matt Jordan)
* ASTERISK-25227 - No audio at in-band announcements in ooh323
channel (Reported by Alexandr Dranchuk)
* ASTERISK-25295 - res_pjsip crash - pjsip_uri_get_uri at
/usr/include/pjsip/sip_uri.h (Reported by Dmitriy Serov)
* ASTERISK-25381 - res_pjsip: AoRs deleted via ARI (or other
mechanism) do not destroy their related contacts (Reported by
Matt Jordan)
* ASTERISK-25369 - res_parking: ParkAndAnnounce - Inheritable
variables aren't applied to the announcer channel (Reported by
Jonathan Rose)
* ASTERISK-25367 - pbx: Long pattern match hints may cause "core
show hints" to crash (Reported by Joshua Colp)
* ASTERISK-25365 - Persistent subscriptions have extra
Content-Length/corrupted messages (Reported by Mark Michelson)
* ASTERISK-25356 - res_pjsip_sdp_rtp: Multiple keepalive scheduled
items may exist (Reported by Joshua Colp)
* ASTERISK-25355 - sched: ast_sched_del may return prematurely due
to spurious wakeup (Reported by Joshua Colp)
* ASTERISK-25318 -
tests/rest_api/applications/subscribe-endpoint/nominal/resource:
Sporadically failing (Reported by Joshua Colp)
* ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
cause on call pickup (Reported by Joshua Colp)
* ASTERISK-25342 - res_pjsip: Repeated usage of pj_gethostip may
block (Reported by Joshua Colp)
* ASTERISK-25341 - bridge: Hangups may get lost when executing
actions (Reported by Joshua Colp)
* ASTERISK-25339 - res_pjsip: Empty "auth" sections from
non-config backgrounds are interpreted as valid (Reported by
Matt Jordan)
* ASTERISK-25215 - Differences in queue.log between Set
QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
Gaetz)
* ASTERISK-25322 - Crash occurs when using MixMonitor with t() or
r() options. (Reported by Richard Mudgett)
* ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
for wrong or non existent peer on invite (Reported by Kevin
Harwell)
* ASTERISK-25312 - res_http_websocket: Terminate connection on
fatal cases (Reported by Joshua Colp)
* ASTERISK-25315 - DAHDI channels send shortened duration DTMF
tones. (Reported by Richard Mudgett)
* ASTERISK-25306 - Persistent subscriptions can save multiple SIP
messages at once, leading to potential crashes. (Reported by
Mark Michelson)
* ASTERISK-25309 - [patch] iLBC 20 advertised (Reported by
Alexander Traud)
* ASTERISK-25304 - res_pjsip: XML sanitization may write past
buffer (Reported by Joshua Colp)
* ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
Firefox 39 - add ECDH support and fallback to prime256v1
(Reported by Stefan Engström)
* ASTERISK-24988 - func_talkdetect: Test is bouncing sporadically
(Reported by Joshua Colp)
* ASTERISK-25181 - ARI: Channels added to Stasis application
during WebSocket creation don't receive a StasisStart event
(Reported by Matt Jordan)
* ASTERISK-25296 - RTP performance issue with several channel
drivers. (Reported by Richard Mudgett)
* ASTERISK-25297 - Crashes running
channels/pjsip/resolver/srv/failover/in_dialog testsuite tests
(Reported by Richard Mudgett)
* ASTERISK-25292 - Testuite:
tests/apps/bridge/bridge_wait/bridge_wait_e_options fails
(Reported by Kevin Harwell)
* ASTERISK-25271 - Parking & blind transfer: Transferer channel
not hung up if no MOH (Reported by Kevin Harwell)
* ASTERISK-25250 - chan_sip - Despite the channel being answered,
caller on a call established via Local channel continues to hear
ringback (Reported by Etienne Lessard)
* ASTERISK-25253 - confbridge volume options and other volume
controls such as func_volume don't work (Reported by Dmitriy
Serov)
* ASTERISK-25247 - choppy audio when spying on a g722 channel,
chan_sip or chan_pjsip (Reported by hristo)
* ASTERISK-25263 - [patch]cdr_adaptive_odbc: CDR insert failure
due to reversed if logic (Reported by Elazar Broad)
* ASTERISK-24867 - Docs for 'e' option in ResetCDR say to use
CDR_PROP instead, CDR_PROP docs are unclear (Reported by Rusty
Newton)
* ASTERISK-24853 - Documentation claims chan_sip outbound
registrations support WS or WSS as valid transports (not true)
(Reported by PSDK)
* ASTERISK-25242 - PJSIP: No audio when Asterisk inside NAT and
endpoints outside NAT - implement functionality similar to
chan_sip 'rtpkeepalive'? (Reported by Mark Michelson)
* ASTERISK-25258 - chan_pjsip: Incorrect format switch on received
RTP packet (Reported by Joshua Colp)
* ASTERISK-25257 - [patch]channels/sig_pri.h -> sig_pri_span ->
force_restart_unavailable_chans in wrong scope (Reported by
Patric Marschall)
* ASTERISK-24934 - [patch]Asterisk manager output does not escape
control characters (Reported by warren smith)
* ASTERISK-25255 - Missing AMI VarSet events when setting to an
empty string. (Reported by Richard Mudgett)
* ASTERISK-25254 - Crash if dialplan sets ATTENDEDTRANSFER to an
empty string before Park. (Reported by Richard Mudgett)
* ASTERISK-25183 - PJSIP: Crash on NULL channel in
chan_pjsip_incoming_response despite previous checks for NULL
channel (Reported by Matt Jordan)
* ASTERISK-25201 - Crash in PJSIP distributor on already free'd
threadpool (Reported by Matt Jordan)
* ASTERISK-25240 - bridge_native_rtp: Direct media wrongfully
started when completing attended transfer (Reported by Joshua
Colp)
* ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
(Reported by Rusty Newton)
* ASTERISK-25146 - DNS: Create system level resolver (Reported by
Joshua Colp)
* ASTERISK-22805 - res_rtp_asterisk: Crash when calling
BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
(Reported by Dmitry Burilov)
* ASTERISK-24550 - res_rtp_asterisk: Crash in
ast_rtp_on_ice_complete during DTLS handshake (Reported by
Osaulenko Alexander)
* ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
Badalian Vyacheslav)
* ASTERISK-24832 - [patch]DTLS-crashes within openssl (Reported
by Stefan Engström)
* ASTERISK-25127 - DTLS crashes following "Unable to cancel
schedule ID" in dtls_srtp_check_pending (Reported by Dade
Brandon)
* ASTERISK-25168 - Random Core Dumps on Asterisk 13.4 PJSIP, in
ast_channel_name at channel_internal_api.c (Reported by Carl
Fortin)
* ASTERISK-25076 - res_pjsip: Failover does not occur on
connection-less transport or 503 response (Reported by Joshua
Colp)
* ASTERISK-25226 - chan_sip: Channel leak in branch 13 on early
replaces call pickup (Reported by Walter Doekes)
* ASTERISK-25222 - Crash in recurring cancel callback called from
ast_dns_resolve_cancel on junk pointer (Reported by Matt Jordan)
* ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
(Reported by Walter Doekes)
* ASTERISK-25219 - [patch]Source and destination overlap in memcpy
in rtp_engine.c (Reported by Walter Doekes)
* ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
(Reported by Walter Doekes)
* ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
Bad file descriptor" (Reported by Barry Chern)
* ASTERISK-25202 - Hints extension state broken between 13.3.2 and
13.4 (Reported by cervajs)
* ASTERISK-25196 - res_pjsip_nat: rewrite_contact should not be
applied to Contact header when Record-Route headers are present
(Reported by Mark Michelson)
* ASTERISK-24907 - res_pjsip_outbound_registration: crash during
unload if registration attempts are still occuring (Reported by
Kevin Harwell)
* ASTERISK-25204 - res_pjsip_refer: Duplicated Referred-By or
Replaces headers on outbound INVITEs. (Reported by Mark
Michelson)
* ASTERISK-25189 - AMI: Add Linkedid header to standard channel
snapshot information. (Reported by Richard Mudgett)
* ASTERISK-25171 - Early completion of feature code attended
transfer results in intermittent one-way audio, "ghost ringing"
and robotic sound. (Reported by Rusty Newton)
* ASTERISK-25172 - Crash in channels/sip/sip blind
transfer/caller_refer_only test in
ast_format_cap_append_from_cap during ast_request (Reported by
Matt Jordan)
* ASTERISK-25180 - res_pjsip_mwi: Unsolicited MWI requires reload
(Reported by Joshua Colp)
* ASTERISK-25182 - [patch] on CLI sip reload, new codecs get
appended only (Reported by Alexander Traud)
* ASTERISK-25163 - Deadlock in chan_sip between reload of sip peer
container and MWI Stasis callback (Reported by Dmitriy Serov)
* ASTERISK-25091 - Asterisk REST API - bridge.addChannel crash
asterisk when calling channel hangup while adding to bridge
(Reported by Ilya Trikoz)
* ASTERISK-24900 - Manager event ParkedCallSwap is not documented
(Reported by Rusty Newton)
* ASTERISK-25162 - func_pjsip_aor: Leak of contact in iterator
(Reported by Corey Farrell)
* ASTERISK-25158 - res_pjsip: Add option to use AAL2 packing when
negotiating g.726 (Reported by Kevin Harwell)
* ASTERISK-24344 - CDR_PROP(disable) disables CDR only for first
dialed party (Reported by Janusz Karolak)
* ASTERISK-24443 - CDR fields (dst, dcontext) empty in transfer
call started from Macro (Reported by Arveno Santoro)
* ASTERISK-25154 - [patch]fromtag may need to be updated after
successful call dialog match (Reported by Damian Ivereigh)
* ASTERISK-25156 - chan_pjsip’s CHAN_START cel event lacks the
correct context and exten (Reported by cloos)
* ASTERISK-25157 - bridging: Performing a blonde transfer does not
result in connected line updates (Reported by Joshua Colp)
* ASTERISK-25087 - Asterisk segfault when using Directory
application with alias option and specific mailbox configuration
(Reported by Chet Stevens)
* ASTERISK-25115 - Crash related to func
sip_resolve_invoke_user_callback of res_pjsip/pjsip_resolver.c
(Reported by John Bigelow)
* ASTERISK-25096 - [patch]Segfault when registering over
websockets with PJSIP (in ast_sockaddr_isnull at
/include/asterisk/netsock2.h) (Reported by Josh Kitchens)
* ASTERISK-24963 - ASAN: heap-use-after-free with PJSIP and WSS
(Reported by Badalian Vyacheslav)
* ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
but asterisk doesn't detect it. (Reported by ibercom)
* ASTERISK-25094 - PBX core: Investigate thread safety issues
(Reported by Corey Farrell)
* ASTERISK-25113 - install_prereq in Debian 8 without "standard
system utilities" (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25148 - res_pjsip NULL channel audit (Reported by Mark
Michelson)
* ASTERISK-25131 - chan_pjsip: In-dialog authentication not
handled. (Reported by Richard Mudgett)
* ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
| adpcm | ipc10} (Reported by Badalian Vyacheslav)
* ASTERISK-25100 - asterisk coredump if host has an IPv6 address
that end with ::80 (Reported by Mark Petersen)
* ASTERISK-25122 - Large SIP packet received via pjsip over
websocket crashes Asterisk (Reported by Ivan Poddubny)
* ASTERISK-25121 - Stasis: Fix unsafe use of stasis_unsubscribe in
modules. (Reported by Corey Farrell)
* ASTERISK-25120 - Astobj2: Weakproxy subscriptions should be run
in reverse order. (Reported by Corey Farrell)
* ASTERISK-25105 - res_pjsip: Possible incompatibility between
qualify_timeout and pjproject-2.4 (Reported by George Joseph)
* ASTERISK-25117 - res_mwi_external_ami: Fix manager action
registrations. (Reported by Corey Farrell)
* ASTERISK-25112 - Logger: Configuration settings are not reset to
default during reload. (Reported by Corey Farrell)
* ASTERISK-24983 - IAX deadlock between hangup and scheduled
actions (ex. largrq) (Reported by Y Ateya)
* ASTERISK-24944 - main/audiohook.c change prevents G722 call
recording (Reported by Ronald Raikes)
* ASTERISK-25110 - res_resolver_unbound.c compilation failure:
SIGURG is undeclared in func unbound_resolver_stop (Reported by
John Bigelow)
* ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
or more digits (Reported by Makoto Dei)
* ASTERISK-25086 - [patch]PJSIP crashes if endpoint missing in
Dial() (Reported by snuffy)
* ASTERISK-25089 - res_pjsip_config_wizard: Variable specified in
templates aren't being processed correctly (Reported by George
Joseph)
* ASTERISK-25090 - CLI core show channel truncates cdr variables
(Reported by snuffy)
* ASTERISK-25083 - Message.c: Message channel becomes saturated
with frames leading to spammy log messages (Reported by Jonathan
Rose)
* ASTERISK-25085 - [patch]Potential crash after unload of
func_periodic_hook or test_message (Reported by Corey Farrell)
* ASTERISK-25082 - Asterisk deletes message after doing a playback
of an INBOX message using ast_vm_play when the Old folder is
full for that mailbox. (Reported by Jonathan Rose)
* ASTERISK-21893 - Segfault after call hangup, in
ast_channel_hangupcause_set, at channel_internal_api.c (Reported
by Aleksandr Gordeev)
* ASTERISK-25042 - asterisk.conf options override command-line
options. (Reported by Corey Farrell)
* ASTERISK-25074 - Regression: Recent clang-related change broke
cross compiling of Asterisk (Reported by Sebastian Kemper)
* ASTERISK-24442 - Outgoing call files don't work properly when
set in the future (Reported by tootai)
* ASTERISK-18252 - queue_log mysql time column data format
(Reported by Gareth Blades)
* ASTERISK-25041 - [patch]Broken column type checking in
res_config_mysql addon (Reported by Alexandre Fournier)
* ASTERISK-25057 - res_pjsip_pubsub: Crash in send_notify due to
invalid root pointer in sub_tree (Reported by Matt Jordan)
* ASTERISK-24938 - ARI Snoop Channel results in excessive
escalating CPU usage (Reported by George Ladoff)
* ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
ignore ISDN RESTART requests. (Reported by Richard Mudgett)
* ASTERISK-25003 - Asterisk crashes on attended transfer (using
feature) (Reported by Artem Volodin)
* ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
contain waiting time (Reported by Etienne Lessard)
* ASTERISK-25027 - Build System: Many ARI modules are missing
dependencies. (Reported by Corey Farrell)
* ASTERISK-25061 - pbx_config: Register manager actions with
module version of macro. (Reported by Corey Farrell)
* ASTERISK-24967 - Problem support schema for pgsql on CEL
(Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25025 - Periodic crashes (in
ast_channel_snapshot_create at stasis_channels.c) with Certified
Asterisk 13. (Reported by Chet Stevens)
* ASTERISK-25053 - Unit test category /main/presence missing
trailing slash. (Reported by Corey Farrell)
* ASTERISK-22708 - res_odbc.conf negative_connection_cache option
not respected, failover between DSNs doesn't work (Reported by
JoshE)
* ASTERISK-25054 - Formats interface's cannot be unregistered,
needs to hold modules until shutdown. (Reported by Corey
Farrell)
* ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
(Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25033 - Asterisk 13 (branch head) won't compile without
PJSip (Reported by Peter Whisker)
* ASTERISK-24896 - [patch] Using force black background leads to
colours not being reset (Reported by dant)
* ASTERISK-25048 - Astobj2: Initialization order wrong when both
refdebug and AO2_DEBUG are both enabled. (Reported by Corey
Farrell)
* ASTERISK-19608 - Asterisk-1.8.x starts rejecting calls with
cause code 44 after some time. (Reported by Denis Alberto
Martinez)
* ASTERISK-25037 - res_pjsip_outbound_registration: Potential
crash in off-nominal failure case when sending message (Reported
by Joshua Colp)
* ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
(Reported by Steve Davies)
* ASTERISK-22790 - check_modem_rate() may return incorrect rate
for V.27 (Reported by not here)
* ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
to minrate=2400, then res_fax refuse to load (Reported by David
Brillert)
* ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
which is disallowed in res_fax's check_modem_rate (Reported by
Matt Jordan)
* ASTERISK-25020 - Mismatched response to outgoing REGISTER
request (Reported by Mark Michelson)
* ASTERISK-25028 - Build System: Unneeded defines in
asterisk/buildopts.h (Reported by Corey Farrell)
* ASTERISK-25026 - Git conversion: Non-C files not switched to
ASTERISK_REGISTER_FILE (Reported by Corey Farrell)
* ASTERISK-24996 - chan_pjsip: Creating Channel Causes Asterisk to
Crash When Duplicate AOR Sections Exist in pjsip.conf (Reported
by Ashley Sanders)
* ASTERISK-25018 - pjsip show endpoints crashes asterisk when
qualified aors present (Reported by Ivan Poddubny)
* ASTERISK-24749 - ConfBridge: Wrong language on playing
conf-hasjoin and conf-hasleft when played to bridge (Reported by
Philippe Bolduc)
* ASTERISK-24845 - pjsip send notify not working with Cisco phone
(Reported by Carl Fortin)
* ASTERISK-25004 - Crash in authenticated reinvite after
originated T.38 FAX (Reported by Mark Michelson)
* ASTERISK-24999 - PJSIP crashes with malformed contact line
(Reported by snuffy)
* ASTERISK-24998 - res_corosync: res_corosync tries to load even
if res_corosync.conf is missing (Reported by George Joseph)
* ASTERISK-24997 - Astobj2: Some callers of __adjust_lock do not
pre-check the object (Reported by Corey Farrell)
* ASTERISK-24994 - dns: Query set unit tests are failing due to
race condition (Reported by Joshua Colp)
* ASTERISK-24982 - res_pjsip_mwi: Unsolicited MWI NOTIFY only sent
on mailbox changes (Reported by Joshua Colp)
* ASTERISK-24991 - Check for ao2_alloc failure in
__ast_channel_internal_alloc (Reported by Corey Farrell)
* ASTERISK-24895 - After hangup on the side of the ISDN network no
HangupRequest event comes for the dahdi channel. (Reported by
Andrew Zherdin)
* ASTERISK-24977 - Contacts that don't use qualify are being
marked as unavailable (Reported by George Joseph)
* ASTERISK-24774 - Segfault in ast_context_destroy with
extensions.ael and extensions.conf (Reported by Corey Farrell)
* ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when
channels have multiple native formats (Reported by Matt Jordan)
* ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
to Fail (Reported by Ashley Sanders)
* ASTERISK-24863 - res_pjsip: No endpoint events raised via AMI
when contacts cannot be reached/qualified (Reported by Dmitriy
Serov)
* ASTERISK-24869 - Asterisk segfaults on DAHDI attended transfer
due to application (appl) being NULL on unbridged channel
(Reported by viniciusfontes)
* ASTERISK-24970 - Crash in res_pjsip_pubsub handling of failed
notify (Reported by Scott Griepentrog)
* ASTERISK-13271 - menuselect sets defaults too late (Reported by
John Nemeth)
* ASTERISK-24959 - [patch]CLI command cdr show pgsql status
(Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-20524 - AMI improperly handles lines of exactly 1025
characters (Reported by David M. Lee)
* ASTERISK-24936 - New Feature: AO2 weakproxy objects (Reported by
Corey Farrell)
* ASTERISK-24954 - Git migration: Asterisk version numbers are
incompatible with the Test Suite (Reported by Matt Jordan)
* ASTERISK-17608 - func_aes.so cannot be loaded if res_crypto /
openssl not compiled (Reported by Warren Selby)
* ASTERISK-24928 - [patch]t38_udptl_maxdatagram in pjsip.conf not
honored (Reported by Juergen Spies)
* ASTERISK-24835 - Early Media Not working with Chan SIP and
Asterisk 13 (Reported by Andrew Nagy)
* ASTERISK-21777 - Asterisk tries to transcode video instead of
audio (Reported by Nick Ruggles)
* ASTERISK-24380 - core: Native formats are set to h264 with
certain audio/video codec configuration, resulting in path
translation WARNINGs (Reported by Matt Jordan)
* ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
into account (Reported by Frederic Van Espen)
* ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
short (Reported by Y Ateya)
* ASTERISK-24935 - res_pjsip_phoneprov_provider: Fix leaked
OBJ_MULTIPLE iterator. (Reported by Corey Farrell)
* ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
(Reported by Vadim)
* ASTERISK-24933 - T38 fails negotiation (Reported by Jonathan
Rose)
* ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
byte prefix bug (Reported by Matt Jordan)
* ASTERISK-21211 - chan_iax2 - unprotected access of
iaxs[peer->callno] potentially results in segfault (Reported by
Jaco Kroon)
* ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
(Reported by Christoph Timm)
* ASTERISK-24910 - "timer=no" and "timer=required" settings in
pjsip.conf fail (Reported by Ray Crumrine)
* ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
(Reported by Jeffrey C. Ollie)
* ASTERISK-24914 - Division by zero in file.c when playback of
voicemail with video as h264 (Reported by Marcello Ceschia)
* ASTERISK-24899 - Parking fall-through behavior different in 13
(Reported by Malcolm Davenport)
* ASTERISK-24937 - [patch]res_pjsip_messaging: Messages may be
sent out of order (Reported by Mark Michelson)
* ASTERISK-24920 - Asterisk handles duplicate SIP requests as if
they were each a new request (Reported by Mark Michelson)
* ASTERISK-24781 - PJSIP: Unnecessary 180 Ringing messages sent
with undesireabe consequences. (Reported by Richard Mudgett)
* ASTERISK-24857 - [patch] "timing test", pjsip incoming/outgoing
calls, voicemail prompts and recordings all fail when using the
kqueue timer source on FreeBSD 10.x (Reported by Justin T.
Gibbs)
* ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
detection in ast_malloc (Reported by Timo Teräs)
* ASTERISK-24142 - CCSS: crash during shutdown due to device
lookup in destroyed container (Reported by David Brillert)
* ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
core restart now (Reported by Peter Katzmann)
* ASTERISK-24805 - [patch] - ASAN: Race condition
(heap-use-after-free) on asterisk closing (Reported by Badalian
Vyacheslav)
* ASTERISK-24881 - ast_register_atexit should only be used when
absolutely needed (Reported by Corey Farrell)
* ASTERISK-24731 - res_pjsip_session cannot be unloaded (Reported
by Corey Farrell)
* ASTERISK-24864 - app_confbridge: file playback blocks dtmf
(Reported by Kevin Harwell)
* ASTERISK-14233 - [patch] Buddies are always auto-registered when
processing the roster (Reported by Simon Arlott)
* ASTERISK-24780 - [patch] - Buddies are always auto-registered
when processing the roster (Reported by Simon Arlott)
* ASTERISK-24879 - [patch]Compilation fails due to 64bit time
under OpenBSD (Reported by snuffy)
* ASTERISK-24880 - [patch]Compilation under OpenBSD (Reported by
snuffy)
* ASTERISK-21765 - [patch] - FILE function's length argument
counts from beginning of file rather than the offset (Reported
by John Zhong)
* ASTERISK-24817 - init_logger_chain: unreachable code block
(Reported by Corey Farrell)
* ASTERISK-24882 - chan_sip: Improve usage of REF_DEBUG (Reported
by Corey Farrell)
* ASTERISK-24876 - Investigate reference leaks from
tests/channels/local/local_optimize_away (Reported by Corey
Farrell)
* ASTERISK-24840 - res_pjsip: conflicting endpoint identifiers
(Reported by Kevin Harwell)
* ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
Atis Lezdins)
* ASTERISK-18708 - func_curl hangs channel under load (Reported by
Dave Cabot)
* ASTERISK-21038 - Bad command completion of "core set debug
channel" (Reported by Richard Kenner)
* ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
by Frank DiGennaro)
* ASTERISK-24872 - [patch] AMI PJSIPShowEndpoint closes AMI
connection on error (Reported by Dmitriy Serov)
* ASTERISK-23666 - CLONE - nested functions aren't portable
(Reported by Diederik de Groot)
* ASTERISK-20399 - Compilation on some systems requires the
-fnested-functions flag (Reported by David M. Lee)
* ASTERISK-20850 - [patch]Nested functions aren't portable.
Adapting RAII_VAR to use clang/llvm blocks to get the
same/similar functionality. (Reported by Diederik de Groot)
* ASTERISK-24807 - Missing mandatory field Max-Forwards (Reported
by Anatoli)
* ASTERISK-24808 - res_config_odbc: Improper escaping of
backslashes occurs with MySQL (Reported by Javier Acosta)
* ASTERISK-23390 - NewExten Event with application AGI shows up
before and after AGI runs (Reported by Benjamin Keith Ford)
* ASTERISK-24786 - [patch] - Asterisk terminates when playing a
voicemail stored in LDAP (Reported by Graham Barnett)
* ASTERISK-24739 - [patch] - Out of files -- call fails --
numerous files with inodes from under /usr/share/zoneinfo,
mostly posixrules (Reported by Ed Hynan)
* ASTERISK-24755 - Asterisk sends unexpected early BYE to
transferrer during attended transfer when using a Stasis bridge
(Reported by John Bigelow)
* ASTERISK-24830 - res_rtp_asterisk.c checks USE_PJPROJECT not
HAVE_PJPROJECT (Reported by Stefan Engström)
* ASTERISK-24825 - Caller ID not recognized using
Centrex/Distinctive dialing (Reported by Richard Mudgett)
* ASTERISK-17588 - Caller ID on TDM410P *UK* PSTN (Reported by
Daniel Flounders)
* ASTERISK-24838 - chan_sip: Locking inversion occurs when
building a peer causes a peer poke during request handling
(Reported by Richard Mudgett)
* ASTERISK-24751 - Integer values in json payload to ARI cause
asterisk to crash (Reported by jeffrey putnam)
* ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
* ASTERISK-18105 - most of asterisk modules are unbuildable in
cygwin environment (Reported by feyfre)
* ASTERISK-21845 - maxcalls exceeded, Asterisk sends out 480 and
also BYE (Reported by Tony Ching)
* ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
error response and BYE are sent to the caller (Reported by
Makoto Dei)
* ASTERISK-23214 - chan_sip WARNING message 'We are requesting
SRTP for audio, but they responded without it' is ambiguous and
wrong in some cases (Reported by Rusty Newton)
* ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
fail (Reported by Terry Wilson)
* ASTERISK-20233 - SRTP not working with some devices (Eg
Grandstream gxv3175) - Message "Can't provide secure audio
requested in SDP offer" (Reported by tootai)
* ASTERISK-22748 - SRTP Crypto Offer With Lifetime Not Accepted
(Reported by Alejandro Mejia)
* ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
thread ID being passed to pthread_kill (Reported by JoshE)
* ASTERISK-24812 - ARI: Creating channels through /channels
resource always uses SLIN, which results in unneeded transcoding
(Reported by Matt Jordan)
* ASTERISK-24797 - bridge_softmix: G.729 codec license held
(Reported by Kevin Harwell)
* ASTERISK-24677 - ARI GET variable on channel provides unhelpful
response on non-existent variable (Reported by Joshua Colp)
* ASTERISK-24785 - 'Expires' header missing from 200 OK on
REGISTER (Reported by Ross Beer)
* ASTERISK-24499 - Need more explicit debug when PJSIP dialstring
is invalid (Reported by Rusty Newton)
* ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
(Reported by Ashley Sanders)
* ASTERISK-24796 - Codecs and bucket schema's prevent module
unload (Reported by Corey Farrell)
* ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
OSX with 64 bit integers (Reported by Corey Farrell)
* ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
for playing back messages stored in IMAP - play_message: No
origtime (Reported by Graham Barnett)
* ASTERISK-22670 - Asterisk crashes when processing ISDN AoC
Events (Reported by klaus3000)
* ASTERISK-24689 - Segfault on hangup after outgoing PRI-Euroisdn
call (Reported by Marcel Manz)
* ASTERISK-24740 - [patch]Segmentation fault on aoc-e event
(Reported by Panos Gkikakis)
* ASTERISK-24799 - [patch] make fails with undefined reference to
SSLv3_client_method (Reported by Alexander Traud)
* ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
(Reported by Corey Farrell)
* ASTERISK-24700 - CRASH: NULL channel is being passed to
ast_bridge_transfer_attended() (Reported by Zane Conkle)
* ASTERISK-24791 - Crash in ast_rtcp_write_report (Reported by
JoshE)
* ASTERISK-24085 - Documentation - We should remove or further
document the 'contact' section in pjsip.conf (Reported by Rusty
Newton)
* ASTERISK-24632 - install_prereq script installs pjproject
without IPv6 support (Reported by Rusty Newton)
* ASTERISK-24685 - "pjsip show version" CLI command (Reported by
Joshua Colp)
* ASTERISK-24768 - res_timing_pthread: file descriptor leak
(Reported by Matthias Urlichs)
* ASTERISK-24612 - res_pjsip: No information if a required sorcery
wizard is not loaded (Reported by Joshua Colp)
* ASTERISK-24716 - Improve pjsip log messages for presence
subscription failure (Reported by Rusty Newton)
* ASTERISK-24771 - ${CHANNEL(pjsip)} - segfault (Reported by
Niklas Larsson)
* ASTERISK-24727 - PJSIP: Crash experienced during multi-Asterisk
transfer scenario. (Reported by Mark Michelson)
* ASTERISK-24015 - app_transfer fails with PJSIP channels
(Reported by Private Name)
* ASTERISK-24741 - dtls_handler causes Asterisk to crash (Reported
by Zane Conkle)
* ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
fully disconnect underlying socket, leading to events being
dropped with no additional information (Reported by Matt Jordan)
* ASTERISK-24752 - Crash in bridge_manager_service_req when bridge
is destroyed by ARI during shutdown (Reported by Richard
Mudgett)
* ASTERISK-24772 - ODBC error in realtime sippeers when device
unregisters under MariaDB (Reported by Richard Miller)
* ASTERISK-24479 - Enable REF_DEBUG for module references
(Reported by Corey Farrell)
* ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
res_odbc (Reported by ibercom)
* ASTERISK-24769 - res_pjsip_sdp_rtp: Local ICE candidates leaked
(Reported by Matt Jordan)
* ASTERISK-24748 - res_pjsip: If wizards explicitly configured in
sorcery.conf false ERROR messages may occur (Reported by Joshua
Colp)
* ASTERISK-24616 - Crash in res_format_attr_h264 due to invalid
string copy (Reported by Yura Kocyuba)
* ASTERISK-24737 - When agent not logged in, agent status shows
unavailable, queue status shows agent invalid (Reported by
Richard Mudgett)
* ASTERISK-24635 - PJSIP outbound PUBLISH crashes when no response
is ever received (Reported by Marco Paland)
* ASTERISK-24736 - Memory Leak Fixes (Reported by Mark Michelson)
* ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
Stephan Eisvogel)
* ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
versions (Reported by Jared Biel)
* ASTERISK-24666 - Security Vulnerability: RTP not closed after
sip call using unsupported codec (Reported by Y Ateya)
* ASTERISK-24676 - Security Vulnerability: URL request injection
in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
* ASTERISK-24729 - Outbound registration not occuring on new
registrations after reload. (Reported by Richard Mudgett)
* ASTERISK-24728 - tcptls: Bad file descriptor error when
reloading chan_sip (Reported by Kevin Harwell)
* ASTERISK-24721 - manager: ModuleLoad action incorrectly reports
'module not found' during a Reload operation (Reported by Matt
Jordan)
* ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
by Kevin Harwell)
* ASTERISK-24485 - res_pjsip cannot be unloaded or shutdown
(Reported by Corey Farrell)
* ASTERISK-24719 - ConfBridge recording channels get stuck when
recording started/stopped more than once (Reported by Richard
Mudgett)
* ASTERISK-24723 - confbridge: CLI command 'confbridge list XXXX'
no longer displays user menus (Reported by Matt Jordan)
* ASTERISK-24539 - Compile fails on OSX because of sem_timedwait
in bridge_channel.c (Reported by George Joseph)
* ASTERISK-24544 - Compile fails on OSX Yosemite because of
incorrect detection of htonll and ntohll (Reported by George
Joseph)
* ASTERISK-24231 - crash: CLI execution of realtime destroy
sippeers id 1 causes crash due to NULL name provided to
ast_variable (Reported by Niklas Larsson)
* ASTERISK-24626 - Voicemail passwords not being stored in ARA
(Reported by Paddy Grice)
* ASTERISK-24693 - Investigate and fix memory leaks in Asterisk
(Reported by Kevin Harwell)
* AS