Asterisk Development Team
2016-Jul-27 21:02 UTC
[asterisk-users] Asterisk 14.0.0-beta1 Now Available
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 H??gh)
* 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 S??bastien 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 D??ms??di)
* 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 Mou??ka)
* 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
Dud??s J??zsef)
* 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 Engstr??m)
* 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 Nem??i??)
* 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 Engstr??m)
* 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 Engstr??m)
* 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 Ter??s)
* 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 Engstr??m)
* 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)
* ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
column comparison for 'defaultuser' (Reported by
HZMI8gkCvPpom0tM)
* ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
m() option does not queue an MWI event (Reported by Gareth
Palmer)
* ASTERISK-24673 - outgoing sip registers cannot be removed or
modified without doing restart (or doing module unload
chan_sip.so) (Reported by Stefan Engstr??m)
* ASTERISK-24640 - Registration pending stays forever after sip
reload (Reported by Max Man)
* ASTERISK-24682 - app_dial: Multiple DialEnd events emitted when
MACRO_RESULT or GOSUB_RESULT are an unexpected value (Reported
by Matt Jordan)
* ASTERISK-24560 - Creating a named ARI bridge twice causes a
crash (Reported by Kinsey Moore)
* ASTERISK-24600 - Stuck IAX channels, Asterisk stops responding
to most traffic, potential deadlock (Reported by Jeff Collell)
* ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
32-bit packages on 64-bit hosts (Reported by Ben Klang)
* ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
voicemail is not deleted after review, hangup (Reported by LEI
FU)
* ASTERISK-24615 - When Multiple Transports Exist in pjsip.conf,
Incorrect External Addresses is Used in SIP Packets When
Responding to INVITE (Reported by David Justl)
* ASTERISK-24624 - Transfer to invalid extension results in hung
channel. (Reported by Zane Conkle)
* ASTERISK-24663 - [patch] Unnamed semaphore autoconf check fails
on cross compilation (Reported by abelbeck)
* ASTERISK-24655 - res_pjsip_outbound_publish: Hang on shutdown
while attempting to publish (Reported by Kevin Harwell)
* ASTERISK-23991 - [patch]asterisk.pc file contains a small error
in the CFlags returned (Reported by Diederik de Groot)
* ASTERISK-23850 - Park Application does not respect Return
Context Priority (Reported by Andrew Nagy)
* ASTERISK-24665 - Configure check required for
pjsip_get_dest_info() (Reported by Mark Michelson)
* ASTERISK-24049 - Asterisk Manager Interface: A number of list
type responses aren't using astman_send_listack (Reported by
Jonathan Rose)
* ASTERISK-20744 - [patch] Security event logging does not work
over syslog (Reported by Michael Keuter)
* ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
(Reported by Kristian H??gh)
* ASTERISK-24474 - sip_to_pjsip.py lacks documentation and does
not function (Reported by John Kiniston)
* ASTERISK-24637 - Channel re-enters Stasis() when it should not
(Reported by John Bigelow)
* ASTERISK-24591 - Stasis() side of an ARI originated channel
cannot be Redirected (Reported by Kinsey Moore)
* ASTERISK-24376 - res_pjsip_refer: REFER request for remote
session attempts to direct channel to external_replaces
extension instead of context, without providing for the
Referred-To SIP URI (Reported by Matt Jordan)
* ASTERISK-24513 - Local channel apparently leaked in off-nominal
DTMF attended transfer (Reported by Mark Michelson)
* ASTERISK-24367 - PJSIP: allow all results in failure to send
INVITE (Reported by Scott Griepentrog)
* ASTERISK-24267 - Queue variables associated with
setinterfacevar, setqueueentryvar, setqueuevar are not passed to
local channel (Reported by Mitch Claborn)
* ASTERISK-24641 - Deadlock in Trunk (Reported by Malcolm
Davenport)
* ASTERISK-23841 - DTMF atxfer doesn't set CallerID for the recall
calls to the transferrer. (Reported by Richard Mudgett)
* ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
destination when 'sendrpid=yes' (in proxy environment) (Reported
by Karsten Wemheuer)
* ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
on startup (Reported by Richard Kenner)
* ASTERISK-24566 - Uninit buf in WS write (Reported by Badalian
Vyacheslav)
* ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
level - 'Remote address is null, most likely RTP has been
stopped' (Reported by Rusty Newton)
* ASTERISK-24459 - bridge_native_rtp: Native RTP bridging is
chosen for RTP compatible channels when the DTMF mode is not
compatible (Reported by Yaniv Simhi)
* ASTERISK-24536 - AMI redirect with PJSIP fails to move extra
channel (Reported by Niklas Larsson)
* ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
casts char to unsigned int (Reported by Walter Doekes)
* ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
enabled (Reported by Andreas Steinmetz)
* ASTERISK-22455 - Asterisk 12 on Ubuntu Lucid deadlocks with
DEBUG_THREADS+OPTIONAL_API enabled (Reported by David M. Lee)
* ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
enabled (Reported by Richard Mudgett)
* ASTERISK-24604 - res_rtp_asterisk: Crash during restart due to
race condition in accessing codec in stored ast_frame and codec
core (Reported by Matt Jordan)
* ASTERISK-24563 - Direct Media calls within private network
sometimes get one way audio (Reported by Kevin Harwell)
* ASTERISK-24607 - res_pjsip_session: re-INVITE with declined
media streams results in 488 (Reported by Matt Jordan)
* ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
from JSSIP (Reported by Badalian Vyacheslav)
* ASTERISK-24514 - res_pjsip_outbound_registration: stack overflow
when using non-default sorcery wizard (Reported by Kevin
Harwell)
* ASTERISK-24342 - PJSIP: Qualifying endpoints attempts to do them
all at the same time. (Reported by Richard Mudgett)
* ASTERISK-24556 - Asterisk 13 core dumps when calling from pjsip
extension to another pjsip extension (Reported by Abhay Gupta)
* ASTERISK-24537 - Stasis: StasisStart/StasisEnd events are not
reliably transmitted during transfers (Reported by Matt Jordan)
* ASTERISK-24573 - [patch]Out of sync conversation recording when
divided in multiple recordings (Reported by Nuno Borges)
* ASTERISK-24572 - [patch]App_meetme is loaded without its
defaults when the configuration file is missing (Reported by
Nuno Borges)
* ASTERISK-24516 - [patch]Asterisk segfaults when playing back
voicemail under high concurrency with an IMAP backend (Reported
by David Duncan Ross Palmer)
* ASTERISK-24274 - [patch]Codec Format Is Not Included in the SDP
Media Attributes When SLIN48 Codec Is Used (Reported by Frankie
Chin)
* ASTERISK-24533 - 2 threads created per chan_sip entry (Reported
by xrobau)
* ASTERISK-24542 - [patch]Failure showing codecs via 'core show
channeltype <tech>' (Reported by snuffy)
* ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
allow blocked addresses through (Reported by Matt Jordan)
* ASTERISK-24534 - [patch]Register DB() as escalating to prevent
users from writing to astdb (Reported by Gareth Palmer)
* ASTERISK-24531 - res_pjsip_acl: ACLs not applied on initial
module load (Reported by Matt Jordan)
* ASTERISK-24490 - Security Vulnerability: CONFBRIDGE function's
record_command option allows arbitrary parameters to be passed
to MixMonitor, allowing remote execution of commands (Reported
by Matt Jordan)
* ASTERISK-24528 - res_pjsip_refer: Sending INVITE with Replaces
in-dialog with invalid target causes crash (Reported by Joshua
Colp)
* ASTERISK-24471 - Crash - assert_fail in libc in
pjmedia_sdp_neg_negotiate from /usr/local/lib/libpjmedia.so.2
(Reported by yaron nahum)
* ASTERISK-24535 - stringfields: Fix regression from fix for
unintentional memory retention and another issue exposed by the
fix (Reported by Corey Farrell)
* ASTERISK-24508 - pjsip - REFER request from SNOM is rejected
with "400 bad request" - DEBUG shows "Received a REFER
without a
parseable Refer-To" (Reported by Beppo Mazzucato)
* ASTERISK-15242 - transmit_refer leaks sip_refer structures
(Reported by David Woolley)
* ASTERISK-24522 - ConfBridge: delay occurs between kicking all
endmarked users when last marked user leaves (Reported by Matt
Jordan)
* ASTERISK-23651 - Reloading some modules that are loaded already,
results in 'No such module' before a successful reload (Reported
by Rusty Newton)
* ASTERISK-24336 - PJSIP timer_min_se value under 90 causes crash
(Reported by Leon Rowland)
* ASTERISK-24501 - ARI: Moving a channel between bridges followed
by a hangup can cause an ARI client to not receive an expected
ChannelLeftBridge event before StasisEnd (Reported by Matt
Jordan)
* ASTERISK-24489 - Crash: Asterisk crashes when converting RTCP
packet to JSON for res_hep_rtcp and report blocks are greater
than 1 (Reported by Gregory Malsack)
* ASTERISK-24498 - Segmentation fault in res_hep_rtcp on attended
transfer (Reported by Beppo Mazzucato)
* ASTERISK-24281 - When bridging 2 chan_sip channels, MOH not
removed from on-hold channels and bridge is never destroyed
after hangup. (Reported by Stefan Engstr??m)
* ASTERISK-24444 - PBX: Crash when generating extension for
pattern matching hint (Reported by Leandro Dardini)
* ASTERISK-24502 - Build fails when dev-mode, dont optimize and
coverage are enabled (Reported by Corey Farrell)
* ASTERISK-24505 - manager: http connections leak references
(Reported by Corey Farrell)
* ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
revision r227276 (Reported by Xavier Hienne)
* ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
length exceeds 50 (roughly) national symbols (Reported by
Dmitriy Bubnov)
* ASTERISK-24250 - [patch] Voicemail with multi-recipients To:
header fix (Reported by abelbeck)
* ASTERISK-24504 - chan_console: Fix reference leaks to pvt
(Reported by Corey Farrell)
* ASTERISK-24447 - Bridge DTMF hooks: Audio doesn't pass when
waiting for more matching digits. (Reported by Richard Mudgett)
* ASTERISK-24257 - agent must dial acceptdtmf twice to bridge to
queue caller (Reported by Steve Pitts)
* ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
extra calls to ast_module_unref (Reported by Corey Farrell)
* ASTERISK-24491 - Memory leak in res_hep (Reported by Zane
Conkle)
* ASTERISK-24307 - Unintentional memory retention in stringfields
(Reported by Etienne Lessard)
* ASTERISK-24438 - res_pjsip_multihomed.so blocks Asterisk reload
when DNS settings invalid (Reported by Melissa Shepherd)
* ASTERISK-20127 - [Regression] Config.c config_text_file_load()
unescapes semicolons ("\;" -> ";") turning them
into comments
(corruption) on rewrite of a config file (Reported by George
Joseph)
* ASTERISK-24487 - configuration: sections should be loadable as
template even when not marked (Reported by Scott Griepentrog)
* ASTERISK-24482 - func_talkdetect: Fix stasis message leak in
audiohook callback (Reported by Corey Farrell)
* ASTERISK-24480 - res_http_websockets: Module reference decrease
below zero (Reported by Corey Farrell)
* ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream
leaks (Reported by Corey Farrell)
* ASTERISK-24411 - [patch] Status of outbound registration is not
changed upon unregistering. (Reported by John Bigelow)
* ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled
(Reported by Corey Farrell)
* ASTERISK-24466 - app_queue: fix a couple leaks to struct
call_queue (Reported by Corey Farrell)
* ASTERISK-24465 - audiohooks list leaks reference to formats
(Reported by Corey Farrell)
* ASTERISK-24462 - res_pjsip: Stale qualify statistics after
disablementation (Reported by Kevin Harwell)
* ASTERISK-24190 - IMAP voicemail causes segfault (Reported by
Nick Adams)
* ASTERISK-24304 - asterisk crashing randomly because of unistim
channel (Reported by dhanapathy sathya)
* ASTERISK-21721 - SIP Failed to parse multiple Supported: headers
(Reported by Olle Johansson)
* ASTERISK-24458 - chan_phone fails to build on big endian systems
(Reported by Tzafrir Cohen)
* ASTERISK-24457 - res_fax: fax gateway frames leak (Reported by
Corey Farrell)
* ASTERISK-24453 - manager: acl_change_sub leaks (Reported by
Corey Farrell)
* ASTERISK-24437 - Review implementation of ast_bridge_impart for
leaks and document proper usage (Reported by Scott Griepentrog)
* ASTERISK-24430 - missing letter "p" in word response in
OriginateResponse event documentation (Reported by Dafi Ni)
* ASTERISK-24323 - Bug in documentation AGI STREAM FILE CONTROL
(Reported by Martin Cis??rik)
* ASTERISK-24419 - Incorrect syntax for setting language in
configs/extensions.conf.sample (Reported by Ben Klang)
* ASTERISK-24454 - app_queue: ao2_iterator not destroyed, causing
leak (Reported by Corey Farrell)
* ASTERISK-24455 - func_cdr: CDR_PROP leaks payload (Reported by
Corey Farrell)
* ASTERISK-24435 - Asterisk 13 with TC400P segfault (Reported by
Marian Koniuszko)
* ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of
SSLv3, security fix POODLE (CVE-2014-3566) (Reported by
abelbeck)
* ASTERISK-24122 - Documentaton for res_pjsip option use_avpf
needs to be fixed (Reported by James Van Vleet)
* ASTERISK-24381 - res_pjsip_sdp_rtp: Declined media streams are
interpreted, leading to erroneous 488 rejections (Reported by
Matt Jordan)
* ASTERISK-24063 - [patch]Asterisk does not respect outbound proxy
when sending qualify requests (Reported by Damian Ivereigh)
* ASTERISK-24415 - Missing AMI VarSet events when channels inherit
variables. (Reported by Richard Mudgett)
* ASTERISK-24327 - bridge_native_rtp: Smart bridge operation to
softmix sometimes fails to properly re-INVITE remotely bridged
participants (Reported by Matt Jordan)
* ASTERISK-24426 - CDR Batch mode: size used as time value after
first expire (Reported by Shane Blaser)
* ASTERISK-24312 - SIGABRT when improperly configured realtime
pjsip (Reported by Dafi Ni)
* ASTERISK-23846 - Unistim multilines. Loss of voice after second
call drops (on a second line). (Reported by Rustam Khankishyiev)
* ASTERISK-24413 - parking/parking_tests: Crash due to assertion
in unit tests when MoH is started on channel in holding bridge
(Reported by Matt Jordan)
* ASTERISK-24393 - rtptimeout=0 doesn't disable rtptimeout
(Reported by Dmitry Melekhov)
* ASTERISK-24321 - SIP deadlock when running automated queues
tests (Reported by Steve Pitts)
* ASTERISK-24392 - res_fax: fax gateway sessions leak (Reported by
Corey Farrell)
* ASTERISK-24237 - CDR: FRACK With PJSIP blonde transfer.
(Reported by Richard Mudgett)
* ASTERISK-24394 - CDR: FRACK with PJSIP directed pickup.
(Reported by Richard Mudgett)
* ASTERISK-18923 - res_fax_spandsp usage counter is wrong
(Reported by Grigoriy Puzankin)
* ASTERISK-22791 - asterisk sends Re-INVITE after receiving a BYE
(Reported by not here)
* ASTERISK-13797 - [patch] relax badshell tilde test (Reported by
Tzafrir Cohen)
* ASTERISK-24325 - res_calendar_ews: cannot be used with neon 0.30
(Reported by Tzafrir Cohen)
* ASTERISK-24406 - Some caller ID strings are parsed differently
since 11.13.0 (Reported by Etienne Lessard)
* ASTERISK-24387 - res_pjsip: rport sent from UAS MUST include the
port that the UAC sent the request on (Reported by Matt Jordan)
* ASTERISK-20784 - Failure to receive an ACK to a SIP Re-INVITE
results in a SIP channel leak (Reported by NITESH BANSAL)
* ASTERISK-15879 - [patch] Failure to receive an ACK to a SIP
Re-INVITE results in a SIP channel leak (Reported by Torrey
Searle)
* ASTERISK-24383 - res_rtp_asterisk: Crash if no candidates
received for component (Reported by Kevin Harwell)
* ASTERISK-24011 - [patch]safe_asterisk tries to set ulimit -n too
high on linux systems with lots of RAM (Reported by Michael
Myles)
* ASTERISK-24326 - res_rtp_asterisk: ICE-TCP candidates are
incorrectly attempted (Reported by Joshua Colp)
* ASTERISK-24389 - chan_iax2: Unit test on Bamboo failing
(Reported by Kevin Harwell)
* ASTERISK-24398 - Initialize auth_rejection_permanent on client
state to the configuration parameter value (Reported by Matt
Jordan)
* ASTERISK-24354 - AMI sendMessage closes AMI connection on error
(Reported by Peter Katzmann)
* ASTERISK-24224 - When using Bridge() dialplan application,
surrogate channel appears in list and call count is inflated.
(Reported by Mark Michelson)
* ASTERISK-24370 - res_pjsip/pjsip_options: OPTIONS request sent
to Asterisk with no user in request is always 404'd (Reported by
Matt Jordan)
* ASTERISK-24382 - chan_pjsip: Calling PJSIP_MEDIA_OFFER on a
non-PJSIP channel results in an invalid reference of a channel
pvt and a FRACK (Reported by Matt Jordan)
* ASTERISK-24369 - res_pjsip: Large message on reliable transport
can cause empty messages to be passed from the PJSIP stack up,
causing crashes in multiple locations (Reported by Matt Jordan)
* ASTERISK-24368 - res_pjsip_pubsub: Subscription persistence
causes crash when re-constructing stored subscription (Reported
by Matt Jordan)
* ASTERISK-24378 - Release AMI connections on shutdown (Reported
by Corey Farrell)
* ASTERISK-24384 - chan_motif: format capabilities leak on module
load error (Reported by Corey Farrell)
* ASTERISK-24199 - 'ALL' is specified in pjsip.conf.sample for TLS
cipher but it is not valid (Reported by Joshua Colp)
* ASTERISK-24195 - bridge_native_rtp: Removing mixmonitor from a
native RTP capable smart bridge doesn't cause the bridge to
resume being a native rtp bridge (Reported by Jonathan Rose)
* ASTERISK-24356 - PJSIP: Directed pickup causes deadlock
(Reported by Richard Mudgett)
* ASTERISK-24262 - AMI CoreShowChannel missing several output
fields and event documentation (Reported by Mitch Claborn)
* ASTERISK-23781 - outgoing missing as enum from
contrib/ast-db-manage/config (Reported by Stephen More)
* ASTERISK-24222 - PJSIP: Failed assertions when placing a call
with no allow= specified (Reported by Mark Michelson)
* ASTERISK-24362 - res_hep leaks reference to configuration
(Reported by Corey Farrell)
* ASTERISK-22945 - [patch] Memory leaks in chan_sip.c with
realtime peers (Reported by ibercom)
* ASTERISK-24350 - PJSIP shows commands prints unneeded headers
(Reported by snuffy)
* ASTERISK-20567 - bashism in autosupport (Reported by Tzafrir
Cohen)
* ASTERISK-24357 - [fax] Out of bounds error in update_modem_bits
(Reported by Jeremy Lain??)
* ASTERISK-24348 - Built-in editline tab complete segfault with
MALLOC_DEBUG (Reported by Walter Doekes)
* ASTERISK-23768 - [patch] Asterisk man page contains a (new)
unquoted minus sign (Reported by Jeremy Lain??)
* ASTERISK-24295 - crash: creating out of dialog OPTIONS request
crashes (Reported by Rogger Padilla)
* ASTERISK-24335 - [PATCH] Asterisk incorrectly responds 503 to
INVITE retransmissions of rejected calls (Reported by Torrey
Searle)
* ASTERISK-24339 - Swagger API Docs have incorrect basePath
(Reported by Bradley Watkins)
* ASTERISK-24265 - segfault in asterisk when try to make call to
IAX (Reported by Dafi Ni)
* ASTERISK-24290 - Endpoint identifier match value fails to parse
when CIDR network format is specified (Reported by Ray Crumrine)
* ASTERISK-24301 - Security: Out of call MESSAGE requests
processed via Message channel driver can crash Asterisk
(Reported by Matt Jordan)
* ASTERISK-24136 - Security: Crash in Asterisk's PJSIP code when
subscribing to an event with an unexpected body type (Reported
by Mark Michelson)
* ASTERISK-24161 - PJSIPShowEndpoint gives inaccurate count of
list items (Reported by Mark Michelson)
* ASTERISK-24331 - Unexpected Errors in Asterisk Manager Interface
Output (Reported by xrobau)
* ASTERISK-24328 - Use of MixMonitor 'm' option results in 0
duration vm description file (Reported by Scott Griepentrog)
* ASTERISK-23577 - res_rtp_asterisk: Crash in
ast_rtp_on_turn_rtp_state when RTP instance is NULL (Reported by
Jay Jideliov)
* ASTERISK-23634 - With TURN Asterisk crashes on multiple (7-10)
concurrent WebRTC (avpg/encryption/icesupport) calls (Reported
by Roman Skvirsky)
* ASTERISK-24249 - SIP debugs do not stop (Reported by Avinash
Mohod)
* ASTERISK-24181 - RLS: Large lists don't get sent because they
exceed the PJSIP message length limit (Reported by Jonathan
Rose)
* ASTERISK-24254 - CDRs: Application/args/dialplan CEP updated
during dial operation (Reported by Matt Jordan)
* ASTERISK-24241 - crash: CDRs recursively attempt to update Party
B information in a multi-party bridge, overrunning the stack
(Reported by Deepak Singh Rawat)
* ASTERISK-24208 - Channels with CDR Information Remain Active
Even After ConfBrige Is Ended (Reported by Frankie Chin)
* ASTERISK-24223 - Gibberish Call-ID on Local channel on
origination (Reported by Mark Michelson)
* ASTERISK-24271 - Unable to make WebRTC call through chan_PJSIP
nor chan_SIP (Reported by Dafi Ni)
* ASTERISK-24212 - testsuite: Sporadic crash due to assert on
stopping RTP engine (Reported by Matt Jordan)
* ASTERISK-24264 - ARI: Adding a channel to a holding bridge
automatically starts MOH (Reported by Samuel Galarneau)
* ASTERISK-23767 - [patch] Dynamic IAX2 registration stops trying
if ever not able to resolve (Reported by David Herselman)
* ASTERISK-24280 - Add 'rtpbindaddr' setting for chan_sip
(Reported by Paul Belanger)
* ASTERISK-24019 - When a Music On Hold stream starts it restarts
at beginning of file. (Reported by Jason Richards)
* ASTERISK-24143 - pjsip: Outbound call to WebRTC UA fails to
transmit ACK on received 200 OK (Reported by Aleksei Kulakov)
* ASTERISK-23997 - chan_sip: port incorrectly incremented for RTCP
ICE candidates in SDP answer (Reported by Badalian Vyacheslav)
* ASTERISK-24147 - ARI: channel hangup crashes asterisk process
(Reported by Edvin Vidmar)
* ASTERISK-23994 - res_pjsip_sdp_rtp: owner address in SDP may not
be fully qualified domainname (Reported by Private Name)
* ASTERISK-22252 - res_musiconhold cleanup - REF_DEBUG reload
warnings and ref leaks (Reported by Walter Doekes)
* ASTERISK-24178 - [patch]fromdomainport used even if not set
(Reported by Elazar Broad)
* ASTERISK-24229 - ARI: playback of sounds implicitly answers
channel, preventing early media playback (Reported by Matt
Jordan)
* ASTERISK-24245 - gcc 4.1.2 complains of files that do not end
with newlines (Reported by Shaun Ruffell)
* ASTERISK-24246 - Quiet warning about type qualifiers ignored on
function return type (Reported by Shaun Ruffell)
* ASTERISK-24043 - ARI /continue fails to actually continue into
the dialplan (Reported by Krandon Bruse)
* ASTERISK-24215 - testsuite: ARI Live Dangerously test fails due
to wrong response code from Asterisk (Reported by Matt Jordan)
* ASTERISK-24134 - ARI: GET /channels/{channel_id}/variable for
channel in dialplan returns 409 conflict (Reported by Matt
Jordan)
* ASTERISK-24138 - dial: Call forwarding information presented
through AMI/ARI is wrong (Reported by Matt Jordan)
* ASTERISK-24234 - app_meetme: Crash on conference shutdown due to
NULL channel passed to meetme_stasis_generate_msg() (Reported by
Shaun Ruffell)
* ASTERISK-24225 - Dial option z is broken (Reported by
dimitripietro)
* ASTERISK-24032 - Gentoo compilation emits warning:
"_FORTIFY_SOURCE" redefined (Reported by Kilburn)
* ASTERISK-24027 - MixMonitor AMI action called during AGI
execution from bridge feature causes channel to leave AGI has
hung up (Reported by Matt Jordan)
* ASTERISK-24236 - res_hep_rtcp: Module incorrectly depends on
pjsip (Reported by Matt Jordan)
* ASTERISK-23508 - Memory Corruption in
__ast_string_field_ptr_build_va (Reported by Arnd Schmitter)
Improvements made in this release:
-----------------------------------
* ASTERISK-26218 - [patch] iLBC 20 (Reported by Alexander Traud)
* ASTERISK-26190 - [patch] SRTP: Enable AES-256 and AES-GCM.
(Reported by Alexander Traud)
* ASTERISK-26220 - Add support for noreturn function attributes.
(Reported by Corey Farrell)
* ASTERISK-22131 - Update the make dependencies script to pull,
build, and install the correct pjproject (Reported by Matt
Jordan)
* ASTERISK-25471 - [patch]Add subscribe_context to res_pjsip
(Reported by JoshE)
* ASTERISK-26159 - res_hep: enabled by default and information
sent to default address (Reported by Ross Beer)
* ASTERISK-26088 - Investigate heavy memory utilization by
res_pjsip_pubsub (Reported by Richard Mudgett)
* ASTERISK-25578 - [patch] SIP/SDP: No rtpmap for static RTP
payload IDs (Reported by Alexander Traud)
* ASTERISK-26011 - [patch]PJSIP: add "via_addr",
"via_port",
"call_id" to contacts (Reported by Alexei Gradinari)
* ASTERISK-25965 - res_pjsip_outbound_publish: Allow multiple
clients per configuration (Reported by Kevin Harwell)
* ASTERISK-25994 - [patch]res_pjsip: module load priority
(Reported by Alexei Gradinari)
* ASTERISK-25931 - PJSIP: add "reg_server" to contacts. (Reported
by Alexei Gradinari)
* ASTERISK-25835 - Authentication using 'Username' field from
Digest (Reported by Ross Beer)
* ASTERISK-25930 - PJSIP: disable multi domain to improve realtime
performace (Reported by Alexei Gradinari)
* ASTERISK-25865 - Message-Account Missing From PJSIP MWI
(Reported by Ross Beer)
* ASTERISK-25444 - [patch]Music On Hold Warning misleading
(Reported by Conrad de Wet)
* ASTERISK-25846 - Gracefully deal with Absent Stasis Apps
(Reported by Andrew Nagy)
* ASTERISK-25791 - res_pjsip_caller_id: Lack of support for
Anonymous <anonymous at anonymous.invalid> (Reported by Anthony
Messina)
* ASTERISK-25767 - [patch] Add check to configure for sanitizes
(Reported by Badalian Vyacheslav)
* ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
core set (Reported by Rusty Newton)
* ASTERISK-25627 - Easily Preventable Compile Warning (Reported by
Diederik de Groot)
* ASTERISK-25558 - [patch]chan_sip option 'notifyringing' doc fix
and addition of 'notifyringingprio' (Reported by Ward van
Wanrooij)
* ASTERISK-25618 - res_pjsip: Check for readability of TLS files
at startup (Reported by George Joseph)
* ASTERISK-25581 - [patch]Add value reason a pause on CLI
(Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25572 - Endpoints: Add StatsD stats for Asterisk
endpoints (Reported by Matt Jordan)
* ASTERISK-25571 - PJSIP: Add StatsD stats for some common PJSIP
objects (Reported by Matt Jordan)
* ASTERISK-25518 - taskprocessor: Add high water mark (Reported by
Jonathan Rose)
* ASTERISK-25495 - [patch] Prevent old-update packages on
repository Debian systems (Reported by Rodrigo Ramirez
Norambuena)
* ASTERISK-25477 - pjsip show "command" like [criteria] (Reported
by Bryant Zimmerman)
* ASTERISK-24718 - [patch]Add inital support of "sanitize" to
configure (Reported by Badalian Vyacheslav)
* ASTERISK-24870 - ARI: Subscriptions to bridges generally not
super useful (Reported by Matt Jordan)
* ASTERISK-25405 - [patch] CLI: core show fd: add timestamp
(Reported by Alexander Traud)
* ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
* ASTERISK-25256 - [patch]Post AMI VarSet to empty string events
when Asterisk deletes a dialplan variable. (Reported by Richard
Mudgett)
* ASTERISK-25040 - pbx: Improve performance of reloads by making
hint destruction more performant (Reported by Matt Jordan)
* ASTERISK-25067 - Sorcery Caching: Implement a new caching module
(Reported by Matt Jordan)
* ASTERISK-25114 - res_pjsip: Add AMI events for chan_pjsip
contact lifecycle changes (Reported by George Joseph)
* ASTERISK-25072 - res_pjsip_outbound_registration: line
functionality. Additional check for using the request URI
(Reported by Dmitriy Serov)
* ASTERISK-24815 - [patch] Enable TLS Dual-Certificates (ECC+RSA)
(Reported by Alexander Traud)
* ASTERISK-25063 - [patch]add X.509 subject alternative name
support to Asterisk TLS support (Reported by Maciej Szmigiero)
* ASTERISK-25044 - sorcery: Add ability to insert a new wizard
into an object type's list (Reported by George Joseph)
* ASTERISK-24892 - Super Awesome Company sound prompts (Reported
by Rusty Newton)
* ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
Hjelm)
* ASTERISK-25049 - CLI: Enable automatic references to modules
(Reported by Corey Farrell)
* ASTERISK-25056 - Modules: Make ast_module_info->self available
to auxiliary sources. (Reported by Corey Farrell)
* ASTERISK-25045 - vector: Add new capabilities and unit tests
(Reported by George Joseph)
* ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
(Reported by Alexander Traud)
* ASTERISK-24706 - [patch]add auto-dtmf mode for pjsip (Reported
by yaron nahum)
* ASTERISK-24917 - [patch] clang compilation warnings (Reported by
Diederik de Groot)
* ASTERISK-25051 - Remove unneeded uses of optional_api providers.
(Reported by Corey Farrell)
* ASTERISK-24974 - Astobj2: Allow reference debugging to be
enabled/disabled by config. (Reported by Corey Farrell)
* ASTERISK-24980 - cdr_adaptive_odbc: refactor lines to
concatenate of columns name (Reported by Rodrigo Ramirez
Norambuena)
* ASTERISK-24947 - res_pjsip: Add a PJSIP resolver using core DNS
(Reported by Joshua Colp)
* ASTERISK-24965 - cel_pgsql - log_error string references CDR
instead of CEL (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-24960 - Build System: Create MOD_ADD_SOURCE macro for
module Makefiles (Reported by Corey Farrell)
* ASTERISK-24939 - [patch]IAX make calltoken expiration time
configurable (Reported by Y Ateya)
* ASTERISK-24918 - pjsip: add CLI options to display global and
system configuration (Reported by Scott Griepentrog)
* ASTERISK-24862 - [patch] Support in-dialog OPTIONS (Reported by
yaron nahum)
* ASTERISK-24802 - stasis: set a channel variable on websocket
disconnect error (Reported by Kevin Harwell)
* ASTERISK-24133 - [patch]Please support Clang; Allow no-exec
stacks (Reported by Jeffrey Walton)
* ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
Couldn't find mailbox %s in context (Reported by Graham Barnett)
* ASTERISK-24813 - asterisk.c: #if statement in listener()
confuses code folding editors (Reported by Corey Farrell)
* ASTERISK-24811 - asterisk-publication sorcery object does not
use realtime (Reported by Matt Hoskins)
* ASTERISK-24745 - [patch]Add no_answer to ARI hangup causes
(Reported by Ben Merrills)
* ASTERISK-24316 - For httpd server, need option to define server
name for security purposes (Reported by Andrew Nagy)
* ASTERISK-24671 - Missing docs for the CDR AMI Event (Reported by
Dan Jenkins)
* ASTERISK-24575 - [patch]Make capath work for res_pjsip (Reported
by cloos)
* ASTERISK-24678 - [PATCH] Added atxfer* settings to
features.conf.sample (Reported by Niklas Larsson)
* ASTERISK-24412 - [patch]Incomplete channel originate/continue
handling with ARI (Reported by Nir Simionovich (GreenfieldTech -
Israel))
* ASTERISK-24351 - [patch] Allow passing options and command to
MixMonitor when recording in ConfBridge (Reported by Gareth
Palmer)
* ASTERISK-24553 - ARI/AMI: Include language in standard channel
snapshot output (Reported by Matt Jordan)
* ASTERISK-24552 - ARI: Allow associating a channel as an
initiator of an Origination for record keeping purposes
(Reported by Matt Jordan)
* ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
lookups (Reported by Birger "WIMPy" Harzenetter)
* ASTERISK-24530 - [patch] app_record stripping 1/4 second from
recordings (Reported by Ben Smithurst)
* ASTERISK-24283 - [patch]Microseconds precision in the eventtime
column in the cel_odbc module (Reported by Etienne Lessard)
* ASTERISK-24128 - [Patch] Adding default dtls settings (Reported
by Michael K.)
* ASTERISK-24279 - Documentation: Clarify the behaviour of the CDR
property 'unanswered' (Reported by Matt Jordan)
* ASTERISK-23512 - Inaccurate comment in manager.conf.sample
(Reported by Richard Miller)
* ASTERISK-24365 - [Patch] Dialplan function to get first/head
caller channel on queue (Reported by Kristian H??gh)
* ASTERISK-23324 - [patch] - QLOOG commiting Japanese translated
prompts (Reported by Kevin McCoy)
* ASTERISK-24038 - device state: Report ONHOLD device state if
channel driver defers device state calculation to core (Reported
by Matt Jordan)
* ASTERISK-24171 - [patch] Provide a manpage for the aelparse
utility (Reported by Jeremy Lain??)
* ASTERISK-23953 - Testsuite: Off-nominal Authenticate test
(Reported by Matt Jordan)
* ASTERISK-24045 - [patch]Voicemail to email at multiple email
addresses (Reported by Jacob Barber)
For a full list of changes in this beta, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.0.0-beta1
Thank you for your continued support of Asterisk!
I'm trying to compile it with unbound but I'm getting the following
error:
"The UNBOUND installation appears to be missing or broken."
Ubuntu 14.04.5 LTS \n \l
root at rtc:/usr/local/src/asterisk-14.0.0-beta1# dpkg -l | grep -i unboun
ii libunbound-dev:amd64 1.4.22-1ubuntu4.14.04.2
amd64 static library, header files, and docs for
libunbound
ii libunbound2:amd64 1.4.22-1ubuntu4.14.04.2
amd64 library implementing DNS resolution and
validation
Any ideas?
Regards,
Marcelo H. Terres <mhterres at gmail.com>
IM: mhterres at jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
On Wed, Jul 27, 2016 at 6:02 PM, Asterisk Development Team
<asteriskteam at digium.com> wrote:> 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 H?gh)
> * 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 S?bastien 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 D?ms?di)
> * 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 Mou?ka)
> * 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
> Dud?s J?zsef)
> * 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 Engstr?m)
> * 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 Nem?i?)
> * 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 Engstr?m)
> * 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 Engstr?m)
> * 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 Ter?s)
> * 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 Engstr?m)
> * 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)
> * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
> column comparison for 'defaultuser' (Reported by
> HZMI8gkCvPpom0tM)
> * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
> m() option does not queue an MWI event (Reported by Gareth
> Palmer)
> * ASTERISK-24673 - outgoing sip registers cannot be removed or
> modified without doing restart (or doing module unload
> chan_sip.so) (Reported by Stefan Engstr?m)
> * ASTERISK-24640 - Registration pending stays forever after sip
> reload (Reported by Max Man)
> * ASTERISK-24682 - app_dial: Multiple DialEnd events emitted when
> MACRO_RESULT or GOSUB_RESULT are an unexpected value (Reported
> by Matt Jordan)
> * ASTERISK-24560 - Creating a named ARI bridge twice causes a
> crash (Reported by Kinsey Moore)
> * ASTERISK-24600 - Stuck IAX channels, Asterisk stops responding
> to most traffic, potential deadlock (Reported by Jeff Collell)
> * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
> 32-bit packages on 64-bit hosts (Reported by Ben Klang)
> * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
> voicemail is not deleted after review, hangup (Reported by LEI
> FU)
> * ASTERISK-24615 - When Multiple Transports Exist in pjsip.conf,
> Incorrect External Addresses is Used in SIP Packets When
> Responding to INVITE (Reported by David Justl)
> * ASTERISK-24624 - Transfer to invalid extension results in hung
> channel. (Reported by Zane Conkle)
> * ASTERISK-24663 - [patch] Unnamed semaphore autoconf check fails
> on cross compilation (Reported by abelbeck)
> * ASTERISK-24655 - res_pjsip_outbound_publish: Hang on shutdown
> while attempting to publish (Reported by Kevin Harwell)
> * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
> in the CFlags returned (Reported by Diederik de Groot)
> * ASTERISK-23850 - Park Application does not respect Return
> Context Priority (Reported by Andrew Nagy)
> * ASTERISK-24665 - Configure check required for
> pjsip_get_dest_info() (Reported by Mark Michelson)
> * ASTERISK-24049 - Asterisk Manager Interface: A number of list
> type responses aren't using astman_send_listack (Reported by
> Jonathan Rose)
> * ASTERISK-20744 - [patch] Security event logging does not work
> over syslog (Reported by Michael Keuter)
> * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
> (Reported by Kristian H?gh)
> * ASTERISK-24474 - sip_to_pjsip.py lacks documentation and does
> not function (Reported by John Kiniston)
> * ASTERISK-24637 - Channel re-enters Stasis() when it should not
> (Reported by John Bigelow)
> * ASTERISK-24591 - Stasis() side of an ARI originated channel
> cannot be Redirected (Reported by Kinsey Moore)
> * ASTERISK-24376 - res_pjsip_refer: REFER request for remote
> session attempts to direct channel to external_replaces
> extension instead of context, without providing for the
> Referred-To SIP URI (Reported by Matt Jordan)
> * ASTERISK-24513 - Local channel apparently leaked in off-nominal
> DTMF attended transfer (Reported by Mark Michelson)
> * ASTERISK-24367 - PJSIP: allow all results in failure to send
> INVITE (Reported by Scott Griepentrog)
> * ASTERISK-24267 - Queue variables associated with
> setinterfacevar, setqueueentryvar, setqueuevar are not passed to
> local channel (Reported by Mitch Claborn)
> * ASTERISK-24641 - Deadlock in Trunk (Reported by Malcolm
> Davenport)
> * ASTERISK-23841 - DTMF atxfer doesn't set CallerID for the recall
> calls to the transferrer. (Reported by Richard Mudgett)
> * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
> destination when 'sendrpid=yes' (in proxy environment)
(Reported
> by Karsten Wemheuer)
> * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
> on startup (Reported by Richard Kenner)
> * ASTERISK-24566 - Uninit buf in WS write (Reported by Badalian
> Vyacheslav)
> * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
> level - 'Remote address is null, most likely RTP has been
> stopped' (Reported by Rusty Newton)
> * ASTERISK-24459 - bridge_native_rtp: Native RTP bridging is
> chosen for RTP compatible channels when the DTMF mode is not
> compatible (Reported by Yaniv Simhi)
> * ASTERISK-24536 - AMI redirect with PJSIP fails to move extra
> channel (Reported by Niklas Larsson)
> * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
> casts char to unsigned int (Reported by Walter Doekes)
> * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
> enabled (Reported by Andreas Steinmetz)
> * ASTERISK-22455 - Asterisk 12 on Ubuntu Lucid deadlocks with
> DEBUG_THREADS+OPTIONAL_API enabled (Reported by David M. Lee)
> * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
> enabled (Reported by Richard Mudgett)
> * ASTERISK-24604 - res_rtp_asterisk: Crash during restart due to
> race condition in accessing codec in stored ast_frame and codec
> core (Reported by Matt Jordan)
> * ASTERISK-24563 - Direct Media calls within private network
> sometimes get one way audio (Reported by Kevin Harwell)
> * ASTERISK-24607 - res_pjsip_session: re-INVITE with declined
> media streams results in 488 (Reported by Matt Jordan)
> * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
> from JSSIP (Reported by Badalian Vyacheslav)
> * ASTERISK-24514 - res_pjsip_outbound_registration: stack overflow
> when using non-default sorcery wizard (Reported by Kevin
> Harwell)
> * ASTERISK-24342 - PJSIP: Qualifying endpoints attempts to do them
> all at the same time. (Reported by Richard Mudgett)
> * ASTERISK-24556 - Asterisk 13 core dumps when calling from pjsip
> extension to another pjsip extension (Reported by Abhay Gupta)
> * ASTERISK-24537 - Stasis: StasisStart/StasisEnd events are not
> reliably transmitted during transfers (Reported by Matt Jordan)
> * ASTERISK-24573 - [patch]Out of sync conversation recording when
> divided in multiple recordings (Reported by Nuno Borges)
> * ASTERISK-24572 - [patch]App_meetme is loaded without its
> defaults when the configuration file is missing (Reported by
> Nuno Borges)
> * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
> voicemail under high concurrency with an IMAP backend (Reported
> by David Duncan Ross Palmer)
> * ASTERISK-24274 - [patch]Codec Format Is Not Included in the SDP
> Media Attributes When SLIN48 Codec Is Used (Reported by Frankie
> Chin)
> * ASTERISK-24533 - 2 threads created per chan_sip entry (Reported
> by xrobau)
> * ASTERISK-24542 - [patch]Failure showing codecs via 'core show
> channeltype <tech>' (Reported by snuffy)
> * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
> allow blocked addresses through (Reported by Matt Jordan)
> * ASTERISK-24534 - [patch]Register DB() as escalating to prevent
> users from writing to astdb (Reported by Gareth Palmer)
> * ASTERISK-24531 - res_pjsip_acl: ACLs not applied on initial
> module load (Reported by Matt Jordan)
> * ASTERISK-24490 - Security Vulnerability: CONFBRIDGE function's
> record_command option allows arbitrary parameters to be passed
> to MixMonitor, allowing remote execution of commands (Reported
> by Matt Jordan)
> * ASTERISK-24528 - res_pjsip_refer: Sending INVITE with Replaces
> in-dialog with invalid target causes crash (Reported by Joshua
> Colp)
> * ASTERISK-24471 - Crash - assert_fail in libc in
> pjmedia_sdp_neg_negotiate from /usr/local/lib/libpjmedia.so.2
> (Reported by yaron nahum)
> * ASTERISK-24535 - stringfields: Fix regression from fix for
> unintentional memory retention and another issue exposed by the
> fix (Reported by Corey Farrell)
> * ASTERISK-24508 - pjsip - REFER request from SNOM is rejected
> with "400 bad request" - DEBUG shows "Received a REFER
without a
> parseable Refer-To" (Reported by Beppo Mazzucato)
> * ASTERISK-15242 - transmit_refer leaks sip_refer structures
> (Reported by David Woolley)
> * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
> endmarked users when last marked user leaves (Reported by Matt
> Jordan)
> * ASTERISK-23651 - Reloading some modules that are loaded already,
> results in 'No such module' before a successful reload
(Reported
> by Rusty Newton)
> * ASTERISK-24336 - PJSIP timer_min_se value under 90 causes crash
> (Reported by Leon Rowland)
> * ASTERISK-24501 - ARI: Moving a channel between bridges followed
> by a hangup can cause an ARI client to not receive an expected
> ChannelLeftBridge event before StasisEnd (Reported by Matt
> Jordan)
> * ASTERISK-24489 - Crash: Asterisk crashes when converting RTCP
> packet to JSON for res_hep_rtcp and report blocks are greater
> than 1 (Reported by Gregory Malsack)
> * ASTERISK-24498 - Segmentation fault in res_hep_rtcp on attended
> transfer (Reported by Beppo Mazzucato)
> * ASTERISK-24281 - When bridging 2 chan_sip channels, MOH not
> removed from on-hold channels and bridge is never destroyed
> after hangup. (Reported by Stefan Engstr?m)
> * ASTERISK-24444 - PBX: Crash when generating extension for
> pattern matching hint (Reported by Leandro Dardini)
> * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
> coverage are enabled (Reported by Corey Farrell)
> * ASTERISK-24505 - manager: http connections leak references
> (Reported by Corey Farrell)
> * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
> revision r227276 (Reported by Xavier Hienne)
> * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
> length exceeds 50 (roughly) national symbols (Reported by
> Dmitriy Bubnov)
> * ASTERISK-24250 - [patch] Voicemail with multi-recipients To:
> header fix (Reported by abelbeck)
> * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
> (Reported by Corey Farrell)
> * ASTERISK-24447 - Bridge DTMF hooks: Audio doesn't pass when
> waiting for more matching digits. (Reported by Richard Mudgett)
> * ASTERISK-24257 - agent must dial acceptdtmf twice to bridge to
> queue caller (Reported by Steve Pitts)
> * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
> extra calls to ast_module_unref (Reported by Corey Farrell)
> * ASTERISK-24491 - Memory leak in res_hep (Reported by Zane
> Conkle)
> * ASTERISK-24307 - Unintentional memory retention in stringfields
> (Reported by Etienne Lessard)
> * ASTERISK-24438 - res_pjsip_multihomed.so blocks Asterisk reload
> when DNS settings invalid (Reported by Melissa Shepherd)
> * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
> unescapes semicolons ("\;" -> ";") turning
them into comments
> (corruption) on rewrite of a config file (Reported by George
> Joseph)
> * ASTERISK-24487 - configuration: sections should be loadable as
> template even when not marked (Reported by Scott Griepentrog)
> * ASTERISK-24482 - func_talkdetect: Fix stasis message leak in
> audiohook callback (Reported by Corey Farrell)
> * ASTERISK-24480 - res_http_websockets: Module reference decrease
> below zero (Reported by Corey Farrell)
> * ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream
> leaks (Reported by Corey Farrell)
> * ASTERISK-24411 - [patch] Status of outbound registration is not
> changed upon unregistering. (Reported by John Bigelow)
> * ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled
> (Reported by Corey Farrell)
> * ASTERISK-24466 - app_queue: fix a couple leaks to struct
> call_queue (Reported by Corey Farrell)
> * ASTERISK-24465 - audiohooks list leaks reference to formats
> (Reported by Corey Farrell)
> * ASTERISK-24462 - res_pjsip: Stale qualify statistics after
> disablementation (Reported by Kevin Harwell)
> * ASTERISK-24190 - IMAP voicemail causes segfault (Reported by
> Nick Adams)
> * ASTERISK-24304 - asterisk crashing randomly because of unistim
> channel (Reported by dhanapathy sathya)
> * ASTERISK-21721 - SIP Failed to parse multiple Supported: headers
> (Reported by Olle Johansson)
> * ASTERISK-24458 - chan_phone fails to build on big endian systems
> (Reported by Tzafrir Cohen)
> * ASTERISK-24457 - res_fax: fax gateway frames leak (Reported by
> Corey Farrell)
> * ASTERISK-24453 - manager: acl_change_sub leaks (Reported by
> Corey Farrell)
> * ASTERISK-24437 - Review implementation of ast_bridge_impart for
> leaks and document proper usage (Reported by Scott Griepentrog)
> * ASTERISK-24430 - missing letter "p" in word response in
> OriginateResponse event documentation (Reported by Dafi Ni)
> * ASTERISK-24323 - Bug in documentation AGI STREAM FILE CONTROL
> (Reported by Martin Cis?rik)
> * ASTERISK-24419 - Incorrect syntax for setting language in
> configs/extensions.conf.sample (Reported by Ben Klang)
> * ASTERISK-24454 - app_queue: ao2_iterator not destroyed, causing
> leak (Reported by Corey Farrell)
> * ASTERISK-24455 - func_cdr: CDR_PROP leaks payload (Reported by
> Corey Farrell)
> * ASTERISK-24435 - Asterisk 13 with TC400P segfault (Reported by
> Marian Koniuszko)
> * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of
> SSLv3, security fix POODLE (CVE-2014-3566) (Reported by
> abelbeck)
> * ASTERISK-24122 - Documentaton for res_pjsip option use_avpf
> needs to be fixed (Reported by James Van Vleet)
> * ASTERISK-24381 - res_pjsip_sdp_rtp: Declined media streams are
> interpreted, leading to erroneous 488 rejections (Reported by
> Matt Jordan)
> * ASTERISK-24063 - [patch]Asterisk does not respect outbound proxy
> when sending qualify requests (Reported by Damian Ivereigh)
> * ASTERISK-24415 - Missing AMI VarSet events when channels inherit
> variables. (Reported by Richard Mudgett)
> * ASTERISK-24327 - bridge_native_rtp: Smart bridge operation to
> softmix sometimes fails to properly re-INVITE remotely bridged
> participants (Reported by Matt Jordan)
> * ASTERISK-24426 - CDR Batch mode: size used as time value after
> first expire (Reported by Shane Blaser)
> * ASTERISK-24312 - SIGABRT when improperly configured realtime
> pjsip (Reported by Dafi Ni)
> * ASTERISK-23846 - Unistim multilines. Loss of voice after second
> call drops (on a second line). (Reported by Rustam Khankishyiev)
> * ASTERISK-24413 - parking/parking_tests: Crash due to assertion
> in unit tests when MoH is started on channel in holding bridge
> (Reported by Matt Jordan)
> * ASTERISK-24393 - rtptimeout=0 doesn't disable rtptimeout
> (Reported by Dmitry Melekhov)
> * ASTERISK-24321 - SIP deadlock when running automated queues
> tests (Reported by Steve Pitts)
> * ASTERISK-24392 - res_fax: fax gateway sessions leak (Reported by
> Corey Farrell)
> * ASTERISK-24237 - CDR: FRACK With PJSIP blonde transfer.
> (Reported by Richard Mudgett)
> * ASTERISK-24394 - CDR: FRACK with PJSIP directed pickup.
> (Reported by Richard Mudgett)
> * ASTERISK-18923 - res_fax_spandsp usage counter is wrong
> (Reported by Grigoriy Puzankin)
> * ASTERISK-22791 - asterisk sends Re-INVITE after receiving a BYE
> (Reported by not here)
> * ASTERISK-13797 - [patch] relax badshell tilde test (Reported by
> Tzafrir Cohen)
> * ASTERISK-24325 - res_calendar_ews: cannot be used with neon 0.30
> (Reported by Tzafrir Cohen)
> * ASTERISK-24406 - Some caller ID strings are parsed differently
> since 11.13.0 (Reported by Etienne Lessard)
> * ASTERISK-24387 - res_pjsip: rport sent from UAS MUST include the
> port that the UAC sent the request on (Reported by Matt Jordan)
> * ASTERISK-20784 - Failure to receive an ACK to a SIP Re-INVITE
> results in a SIP channel leak (Reported by NITESH BANSAL)
> * ASTERISK-15879 - [patch] Failure to receive an ACK to a SIP
> Re-INVITE results in a SIP channel leak (Reported by Torrey
> Searle)
> * ASTERISK-24383 - res_rtp_asterisk: Crash if no candidates
> received for component (Reported by Kevin Harwell)
> * ASTERISK-24011 - [patch]safe_asterisk tries to set ulimit -n too
> high on linux systems with lots of RAM (Reported by Michael
> Myles)
> * ASTERISK-24326 - res_rtp_asterisk: ICE-TCP candidates are
> incorrectly attempted (Reported by Joshua Colp)
> * ASTERISK-24389 - chan_iax2: Unit test on Bamboo failing
> (Reported by Kevin Harwell)
> * ASTERISK-24398 - Initialize auth_rejection_permanent on client
> state to the configuration parameter value (Reported by Matt
> Jordan)
> * ASTERISK-24354 - AMI sendMessage closes AMI connection on error
> (Reported by Peter Katzmann)
> * ASTERISK-24224 - When using Bridge() dialplan application,
> surrogate channel appears in list and call count is inflated.
> (Reported by Mark Michelson)
> * ASTERISK-24370 - res_pjsip/pjsip_options: OPTIONS request sent
> to Asterisk with no user in request is always 404'd (Reported by
> Matt Jordan)
> * ASTERISK-24382 - chan_pjsip: Calling PJSIP_MEDIA_OFFER on a
> non-PJSIP channel results in an invalid reference of a channel
> pvt and a FRACK (Reported by Matt Jordan)
> * ASTERISK-24369 - res_pjsip: Large message on reliable transport
> can cause empty messages to be passed from the PJSIP stack up,
> causing crashes in multiple locations (Reported by Matt Jordan)
> * ASTERISK-24368 - res_pjsip_pubsub: Subscription persistence
> causes crash when re-constructing stored subscription (Reported
> by Matt Jordan)
> * ASTERISK-24378 - Release AMI connections on shutdown (Reported
> by Corey Farrell)
> * ASTERISK-24384 - chan_motif: format capabilities leak on module
> load error (Reported by Corey Farrell)
> * ASTERISK-24199 - 'ALL' is specified in pjsip.conf.sample for TLS
> cipher but it is not valid (Reported by Joshua Colp)
> * ASTERISK-24195 - bridge_native_rtp: Removing mixmonitor from a
> native RTP capable smart bridge doesn't cause the bridge to
> resume being a native rtp bridge (Reported by Jonathan Rose)
> * ASTERISK-24356 - PJSIP: Directed pickup causes deadlock
> (Reported by Richard Mudgett)
> * ASTERISK-24262 - AMI CoreShowChannel missing several output
> fields and event documentation (Reported by Mitch Claborn)
> * ASTERISK-23781 - outgoing missing as enum from
> contrib/ast-db-manage/config (Reported by Stephen More)
> * ASTERISK-24222 - PJSIP: Failed assertions when placing a call
> with no allow= specified (Reported by Mark Michelson)
> * ASTERISK-24362 - res_hep leaks reference to configuration
> (Reported by Corey Farrell)
> * ASTERISK-22945 - [patch] Memory leaks in chan_sip.c with
> realtime peers (Reported by ibercom)
> * ASTERISK-24350 - PJSIP shows commands prints unneeded headers
> (Reported by snuffy)
> * ASTERISK-20567 - bashism in autosupport (Reported by Tzafrir
> Cohen)
> * ASTERISK-24357 - [fax] Out of bounds error in update_modem_bits
> (Reported by Jeremy Lain?)
> * ASTERISK-24348 - Built-in editline tab complete segfault with
> MALLOC_DEBUG (Reported by Walter Doekes)
> * ASTERISK-23768 - [patch] Asterisk man page contains a (new)
> unquoted minus sign (Reported by Jeremy Lain?)
> * ASTERISK-24295 - crash: creating out of dialog OPTIONS request
> crashes (Reported by Rogger Padilla)
> * ASTERISK-24335 - [PATCH] Asterisk incorrectly responds 503 to
> INVITE retransmissions of rejected calls (Reported by Torrey
> Searle)
> * ASTERISK-24339 - Swagger API Docs have incorrect basePath
> (Reported by Bradley Watkins)
> * ASTERISK-24265 - segfault in asterisk when try to make call to
> IAX (Reported by Dafi Ni)
> * ASTERISK-24290 - Endpoint identifier match value fails to parse
> when CIDR network format is specified (Reported by Ray Crumrine)
> * ASTERISK-24301 - Security: Out of call MESSAGE requests
> processed via Message channel driver can crash Asterisk
> (Reported by Matt Jordan)
> * ASTERISK-24136 - Security: Crash in Asterisk's PJSIP code when
> subscribing to an event with an unexpected body type (Reported
> by Mark Michelson)
> * ASTERISK-24161 - PJSIPShowEndpoint gives inaccurate count of
> list items (Reported by Mark Michelson)
> * ASTERISK-24331 - Unexpected Errors in Asterisk Manager Interface
> Output (Reported by xrobau)
> * ASTERISK-24328 - Use of MixMonitor 'm' option results in 0
> duration vm description file (Reported by Scott Griepentrog)
> * ASTERISK-23577 - res_rtp_asterisk: Crash in
> ast_rtp_on_turn_rtp_state when RTP instance is NULL (Reported by
> Jay Jideliov)
> * ASTERISK-23634 - With TURN Asterisk crashes on multiple (7-10)
> concurrent WebRTC (avpg/encryption/icesupport) calls (Reported
> by Roman Skvirsky)
> * ASTERISK-24249 - SIP debugs do not stop (Reported by Avinash
> Mohod)
> * ASTERISK-24181 - RLS: Large lists don't get sent because they
> exceed the PJSIP message length limit (Reported by Jonathan
> Rose)
> * ASTERISK-24254 - CDRs: Application/args/dialplan CEP updated
> during dial operation (Reported by Matt Jordan)
> * ASTERISK-24241 - crash: CDRs recursively attempt to update Party
> B information in a multi-party bridge, overrunning the stack
> (Reported by Deepak Singh Rawat)
> * ASTERISK-24208 - Channels with CDR Information Remain Active
> Even After ConfBrige Is Ended (Reported by Frankie Chin)
> * ASTERISK-24223 - Gibberish Call-ID on Local channel on
> origination (Reported by Mark Michelson)
> * ASTERISK-24271 - Unable to make WebRTC call through chan_PJSIP
> nor chan_SIP (Reported by Dafi Ni)
> * ASTERISK-24212 - testsuite: Sporadic crash due to assert on
> stopping RTP engine (Reported by Matt Jordan)
> * ASTERISK-24264 - ARI: Adding a channel to a holding bridge
> automatically starts MOH (Reported by Samuel Galarneau)
> * ASTERISK-23767 - [patch] Dynamic IAX2 registration stops trying
> if ever not able to resolve (Reported by David Herselman)
> * ASTERISK-24280 - Add 'rtpbindaddr' setting for chan_sip
> (Reported by Paul Belanger)
> * ASTERISK-24019 - When a Music On Hold stream starts it restarts
> at beginning of file. (Reported by Jason Richards)
> * ASTERISK-24143 - pjsip: Outbound call to WebRTC UA fails to
> transmit ACK on received 200 OK (Reported by Aleksei Kulakov)
> * ASTERISK-23997 - chan_sip: port incorrectly incremented for RTCP
> ICE candidates in SDP answer (Reported by Badalian Vyacheslav)
> * ASTERISK-24147 - ARI: channel hangup crashes asterisk process
> (Reported by Edvin Vidmar)
> * ASTERISK-23994 - res_pjsip_sdp_rtp: owner address in SDP may not
> be fully qualified domainname (Reported by Private Name)
> * ASTERISK-22252 - res_musiconhold cleanup - REF_DEBUG reload
> warnings and ref leaks (Reported by Walter Doekes)
> * ASTERISK-24178 - [patch]fromdomainport used even if not set
> (Reported by Elazar Broad)
> * ASTERISK-24229 - ARI: playback of sounds implicitly answers
> channel, preventing early media playback (Reported by Matt
> Jordan)
> * ASTERISK-24245 - gcc 4.1.2 complains of files that do not end
> with newlines (Reported by Shaun Ruffell)
> * ASTERISK-24246 - Quiet warning about type qualifiers ignored on
> function return type (Reported by Shaun Ruffell)
> * ASTERISK-24043 - ARI /continue fails to actually continue into
> the dialplan (Reported by Krandon Bruse)
> * ASTERISK-24215 - testsuite: ARI Live Dangerously test fails due
> to wrong response code from Asterisk (Reported by Matt Jordan)
> * ASTERISK-24134 - ARI: GET /channels/{channel_id}/variable for
> channel in dialplan returns 409 conflict (Reported by Matt
> Jordan)
> * ASTERISK-24138 - dial: Call forwarding information presented
> through AMI/ARI is wrong (Reported by Matt Jordan)
> * ASTERISK-24234 - app_meetme: Crash on conference shutdown due to
> NULL channel passed to meetme_stasis_generate_msg() (Reported by
> Shaun Ruffell)
> * ASTERISK-24225 - Dial option z is broken (Reported by
> dimitripietro)
> * ASTERISK-24032 - Gentoo compilation emits warning:
> "_FORTIFY_SOURCE" redefined (Reported by Kilburn)
> * ASTERISK-24027 - MixMonitor AMI action called during AGI
> execution from bridge feature causes channel to leave AGI has
> hung up (Reported by Matt Jordan)
> * ASTERISK-24236 - res_hep_rtcp: Module incorrectly depends on
> pjsip (Reported by Matt Jordan)
> * ASTERISK-23508 - Memory Corruption in
> __ast_string_field_ptr_build_va (Reported by Arnd Schmitter)
>
> Improvements made in this release:
> -----------------------------------
> * ASTERISK-26218 - [patch] iLBC 20 (Reported by Alexander Traud)
> * ASTERISK-26190 - [patch] SRTP: Enable AES-256 and AES-GCM.
> (Reported by Alexander Traud)
> * ASTERISK-26220 - Add support for noreturn function attributes.
> (Reported by Corey Farrell)
> * ASTERISK-22131 - Update the make dependencies script to pull,
> build, and install the correct pjproject (Reported by Matt
> Jordan)
> * ASTERISK-25471 - [patch]Add subscribe_context to res_pjsip
> (Reported by JoshE)
> * ASTERISK-26159 - res_hep: enabled by default and information
> sent to default address (Reported by Ross Beer)
> * ASTERISK-26088 - Investigate heavy memory utilization by
> res_pjsip_pubsub (Reported by Richard Mudgett)
> * ASTERISK-25578 - [patch] SIP/SDP: No rtpmap for static RTP
> payload IDs (Reported by Alexander Traud)
> * ASTERISK-26011 - [patch]PJSIP: add "via_addr",
"via_port",
> "call_id" to contacts (Reported by Alexei Gradinari)
> * ASTERISK-25965 - res_pjsip_outbound_publish: Allow multiple
> clients per configuration (Reported by Kevin Harwell)
> * ASTERISK-25994 - [patch]res_pjsip: module load priority
> (Reported by Alexei Gradinari)
> * ASTERISK-25931 - PJSIP: add "reg_server" to contacts.
(Reported
> by Alexei Gradinari)
> * ASTERISK-25835 - Authentication using 'Username' field from
> Digest (Reported by Ross Beer)
> * ASTERISK-25930 - PJSIP: disable multi domain to improve realtime
> performace (Reported by Alexei Gradinari)
> * ASTERISK-25865 - Message-Account Missing From PJSIP MWI
> (Reported by Ross Beer)
> * ASTERISK-25444 - [patch]Music On Hold Warning misleading
> (Reported by Conrad de Wet)
> * ASTERISK-25846 - Gracefully deal with Absent Stasis Apps
> (Reported by Andrew Nagy)
> * ASTERISK-25791 - res_pjsip_caller_id: Lack of support for
> Anonymous <anonymous at anonymous.invalid> (Reported by Anthony
> Messina)
> * ASTERISK-25767 - [patch] Add check to configure for sanitizes
> (Reported by Badalian Vyacheslav)
> * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
> core set (Reported by Rusty Newton)
> * ASTERISK-25627 - Easily Preventable Compile Warning (Reported by
> Diederik de Groot)
> * ASTERISK-25558 - [patch]chan_sip option 'notifyringing' doc fix
> and addition of 'notifyringingprio' (Reported by Ward van
> Wanrooij)
> * ASTERISK-25618 - res_pjsip: Check for readability of TLS files
> at startup (Reported by George Joseph)
> * ASTERISK-25581 - [patch]Add value reason a pause on CLI
> (Reported by Rodrigo Ramirez Norambuena)
> * ASTERISK-25572 - Endpoints: Add StatsD stats for Asterisk
> endpoints (Reported by Matt Jordan)
> * ASTERISK-25571 - PJSIP: Add StatsD stats for some common PJSIP
> objects (Reported by Matt Jordan)
> * ASTERISK-25518 - taskprocessor: Add high water mark (Reported by
> Jonathan Rose)
> * ASTERISK-25495 - [patch] Prevent old-update packages on
> repository Debian systems (Reported by Rodrigo Ramirez
> Norambuena)
> * ASTERISK-25477 - pjsip show "command" like [criteria]
(Reported
> by Bryant Zimmerman)
> * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
> configure (Reported by Badalian Vyacheslav)
> * ASTERISK-24870 - ARI: Subscriptions to bridges generally not
> super useful (Reported by Matt Jordan)
> * ASTERISK-25405 - [patch] CLI: core show fd: add timestamp
> (Reported by Alexander Traud)
> * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
> defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
> * ASTERISK-25256 - [patch]Post AMI VarSet to empty string events
> when Asterisk deletes a dialplan variable. (Reported by Richard
> Mudgett)
> * ASTERISK-25040 - pbx: Improve performance of reloads by making
> hint destruction more performant (Reported by Matt Jordan)
> * ASTERISK-25067 - Sorcery Caching: Implement a new caching module
> (Reported by Matt Jordan)
> * ASTERISK-25114 - res_pjsip: Add AMI events for chan_pjsip
> contact lifecycle changes (Reported by George Joseph)
> * ASTERISK-25072 - res_pjsip_outbound_registration: line
> functionality. Additional check for using the request URI
> (Reported by Dmitriy Serov)
> * ASTERISK-24815 - [patch] Enable TLS Dual-Certificates (ECC+RSA)
> (Reported by Alexander Traud)
> * ASTERISK-25063 - [patch]add X.509 subject alternative name
> support to Asterisk TLS support (Reported by Maciej Szmigiero)
> * ASTERISK-25044 - sorcery: Add ability to insert a new wizard
> into an object type's list (Reported by George Joseph)
> * ASTERISK-24892 - Super Awesome Company sound prompts (Reported
> by Rusty Newton)
> * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
> Hjelm)
> * ASTERISK-25049 - CLI: Enable automatic references to modules
> (Reported by Corey Farrell)
> * ASTERISK-25056 - Modules: Make ast_module_info->self available
> to auxiliary sources. (Reported by Corey Farrell)
> * ASTERISK-25045 - vector: Add new capabilities and unit tests
> (Reported by George Joseph)
> * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
> (Reported by Alexander Traud)
> * ASTERISK-24706 - [patch]add auto-dtmf mode for pjsip (Reported
> by yaron nahum)
> * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
> Diederik de Groot)
> * ASTERISK-25051 - Remove unneeded uses of optional_api providers.
> (Reported by Corey Farrell)
> * ASTERISK-24974 - Astobj2: Allow reference debugging to be
> enabled/disabled by config. (Reported by Corey Farrell)
> * ASTERISK-24980 - cdr_adaptive_odbc: refactor lines to
> concatenate of columns name (Reported by Rodrigo Ramirez
> Norambuena)
> * ASTERISK-24947 - res_pjsip: Add a PJSIP resolver using core DNS
> (Reported by Joshua Colp)
> * ASTERISK-24965 - cel_pgsql - log_error string references CDR
> instead of CEL (Reported by Rodrigo Ramirez Norambuena)
> * ASTERISK-24960 - Build System: Create MOD_ADD_SOURCE macro for
> module Makefiles (Reported by Corey Farrell)
> * ASTERISK-24939 - [patch]IAX make calltoken expiration time
> configurable (Reported by Y Ateya)
> * ASTERISK-24918 - pjsip: add CLI options to display global and
> system configuration (Reported by Scott Griepentrog)
> * ASTERISK-24862 - [patch] Support in-dialog OPTIONS (Reported by
> yaron nahum)
> * ASTERISK-24802 - stasis: set a channel variable on websocket
> disconnect error (Reported by Kevin Harwell)
> * ASTERISK-24133 - [patch]Please support Clang; Allow no-exec
> stacks (Reported by Jeffrey Walton)
> * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
> Couldn't find mailbox %s in context (Reported by Graham Barnett)
> * ASTERISK-24813 - asterisk.c: #if statement in listener()
> confuses code folding editors (Reported by Corey Farrell)
> * ASTERISK-24811 - asterisk-publication sorcery object does not
> use realtime (Reported by Matt Hoskins)
> * ASTERISK-24745 - [patch]Add no_answer to ARI hangup causes
> (Reported by Ben Merrills)
> * ASTERISK-24316 - For httpd server, need option to define server
> name for security purposes (Reported by Andrew Nagy)
> * ASTERISK-24671 - Missing docs for the CDR AMI Event (Reported by
> Dan Jenkins)
> * ASTERISK-24575 - [patch]Make capath work for res_pjsip (Reported
> by cloos)
> * ASTERISK-24678 - [PATCH] Added atxfer* settings to
> features.conf.sample (Reported by Niklas Larsson)
> * ASTERISK-24412 - [patch]Incomplete channel originate/continue
> handling with ARI (Reported by Nir Simionovich (GreenfieldTech -
> Israel))
> * ASTERISK-24351 - [patch] Allow passing options and command to
> MixMonitor when recording in ConfBridge (Reported by Gareth
> Palmer)
> * ASTERISK-24553 - ARI/AMI: Include language in standard channel
> snapshot output (Reported by Matt Jordan)
> * ASTERISK-24552 - ARI: Allow associating a channel as an
> initiator of an Origination for record keeping purposes
> (Reported by Matt Jordan)
> * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
> lookups (Reported by Birger "WIMPy" Harzenetter)
> * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
> recordings (Reported by Ben Smithurst)
> * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
> column in the cel_odbc module (Reported by Etienne Lessard)
> * ASTERISK-24128 - [Patch] Adding default dtls settings (Reported
> by Michael K.)
> * ASTERISK-24279 - Documentation: Clarify the behaviour of the CDR
> property 'unanswered' (Reported by Matt Jordan)
> * ASTERISK-23512 - Inaccurate comment in manager.conf.sample
> (Reported by Richard Miller)
> * ASTERISK-24365 - [Patch] Dialplan function to get first/head
> caller channel on queue (Reported by Kristian H?gh)
> * ASTERISK-23324 - [patch] - QLOOG commiting Japanese translated
> prompts (Reported by Kevin McCoy)
> * ASTERISK-24038 - device state: Report ONHOLD device state if
> channel driver defers device state calculation to core (Reported
> by Matt Jordan)
> * ASTERISK-24171 - [patch] Provide a manpage for the aelparse
> utility (Reported by Jeremy Lain?)
> * ASTERISK-23953 - Testsuite: Off-nominal Authenticate test
> (Reported by Matt Jordan)
> * ASTERISK-24045 - [patch]Voicemail to email at multiple email
> addresses (Reported by Jacob Barber)
>
> For a full list of changes in this beta, please see the ChangeLog:
>
> http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.0.0-beta1
>
> Thank you for your continued support of Asterisk!
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
I'm trying to compile it with unbound but I'm getting the following
error:
"The UNBOUND installation appears to be missing or broken."
Ubuntu 14.04.5 LTS \n \l
root at rtc:/usr/local/src/asterisk-14.0.0-beta1# dpkg -l | grep -i unboun
ii libunbound-dev:amd64 1.4.22-1ubuntu4.14.04.2
amd64 static library, header files, and docs for
libunbound
ii libunbound2:amd64 1.4.22-1ubuntu4.14.04.2
amd64 library implementing DNS resolution and
validation
Any ideas?
Marcelo H. Terres <mhterres at gmail.com>
IM: mhterres at jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
On Wed, Jul 27, 2016 at 6:02 PM, Asterisk Development Team
<asteriskteam at digium.com> wrote:> 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!
>
Marcelo Terres wrote:> I'm trying to compile it with unbound but I'm getting the following error: > > "The UNBOUND installation appears to be missing or broken." > > Ubuntu 14.04.5 LTS \n \l > > root at rtc:/usr/local/src/asterisk-14.0.0-beta1# dpkg -l | grep -i unboun > ii libunbound-dev:amd64 1.4.22-1ubuntu4.14.04.2 > amd64 static library, header files, and docs for > libunbound > ii libunbound2:amd64 1.4.22-1ubuntu4.14.04.2 > amd64 library implementing DNS resolution and > validation > > Any ideas?The version and capability check for unbound was too strict and has been tweaked since the initial beta1 release. The next beta (or rc) will have the fix, and it's confirmed to work against Ubuntu 14.04. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org