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<div class="moz-cite-prefix">Hi,<br>
<br>
The configure not tests the libsrtp dependencies.<br>
If you not install it you will see the error :<br>
<br>
sdp_srtp.c:37:72: fatal error: srtp/srtp.h: No such file or
directory<br>
#include <srtp/srtp.h> /* for
SRTP_MAX_KEY_LEN, etc */<br>
<br>
For debian use :<br>
<br>
# apt-get install libsrtp0-dev<br>
<br>
Regards,<br>
<br>
<br>
Borja SIXTO from Ulex.fr<br>
<br>
Le 27/07/2016 à 21:52, Asterisk Development Team a écrit :<br>
</div>
<blockquote cite="mid:E1bSUsd-0000N7-0O@mail.digium.com" type="cite">
<pre wrap="">The Asterisk Development Team has announced the first beta of
Asterisk 14.0.0. This beta is available for immediate
download at <a class="moz-txt-link-freetext" href="http://downloads.asterisk.org/pub/telephony/asterisk">http://downloads.asterisk.org/pub/telephony/asterisk</a>
The release of Asterisk 14.0.0-beta1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this beta:
New Features made in this release:
-----------------------------------
* ASTERISK-25904 - PJSIP: add contact.updated event (Reported by
Alexei Gradinari)
* ASTERISK-26058 - [Patch] Add uptime and last reloaded to
FullyBooted AMI event (Reported by Niklas Larsson)
* ASTERISK-25925 - Allow Early Bridges on ARI Dials (Reported by
Mark Michelson)
* ASTERISK-26068 - Multicast RTP Options (Reported by Mark
Michelson)
* ASTERISK-26042 - ARI: Allow downloading of the media associated
with a stored recording (Reported by Matt Jordan)
* ASTERISK-25425 - logger: Add JSON structured logging (Reported
by Matt Jordan)
* ASTERISK-25900 - PJSIP Endpoint IP Access Controls (Reported by
Alexei Gradinari)
* ASTERISK-25972 - res_pjsip_exten_state: Use body generator to
publish extension state (Reported by Richard Mudgett)
* ASTERISK-25889 - ARI: Add separate "create" and "dial"
operations for channels (Reported by Mark Michelson)
* ASTERISK-25803 - [patch] chan_sip: Optionally supply
fromuser/fromdomain in SIP dial string (Reported by Walter
Doekes)
* ASTERISK-24919 - res_pjsip_config_wizard: Ability to write
contents to file (Reported by Ray Crumrine)
* ASTERISK-25670 - Add regcontext to PJSIP (Reported by Daniel
Journo)
* ASTERISK-25660 - Add sipp-sendfax.xml and spandspflow2pcap.py to
contrib/scripts. (Reported by Walter Doekes)
* ASTERISK-25591 - [patch] Complete List of Header Files
(#include): iwyu (Reported by Alexander Traud)
* ASTERISK-25551 - [patch]Ability to add channel to an existing
bridge by specifying an existing channel prefix (Reported by
Alec Davis)
* ASTERISK-25419 - Dialplan Application for Integration of StatsD
(Reported by Ashley Sanders)
* ASTERISK-25549 - Confbridge: Add participant timeout option
(Reported by Mark Michelson)
* ASTERISK-24922 - ARI: Add the ability to intercept hold and
raise an event (Reported by Matt Jordan)
* ASTERISK-25479 - Allow CDR's to be modified before being
dispatched to engines (Reported by Jonh Wendell)
* ASTERISK-25480 - [patch]Add field PauseReason on
QueueMemberStatus (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25377 - res_pjsip: Change default "From user" from UUID
to something more palatable (Reported by Mark Michelson)
* ASTERISK-25252 - ARI: Add the ability to manipulate log channels
(Reported by Matt Jordan)
* ASTERISK-25259 - chan_pjsip: Add rtptimeout support (Reported by
Joshua Colp)
* ASTERISK-25238 - ARI: Support push configuration (Reported by
Matt Jordan)
* ASTERISK-25173 - ARI: Add the ability to load/reload/unload an
Asterisk module (Reported by Matt Jordan)
* ASTERISK-25006 - [patch] Add support set character for quoted
identifiers (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-23186 - [patch] Add usegmtime option to cel_pgsql
(Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-24931 - dns: Add support for SRV records. (Reported by
Joshua Colp)
* ASTERISK-24834 - DNS Overhaul: Implement the proposed core API -
sync/async functions, resolver registration (Reported by Matt
Jordan)
* ASTERISK-24836 - DNS Overhaul: Write a Resolver Implementation
(Reported by Matt Jordan)
* ASTERISK-22591 - [patch]Prevent Asterisk from writing received
SMS content in log (Reported by Jan Juergens)
* ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
(Reported by Dwayne Hubbard)
* ASTERISK-24703 - ARI: Add the ability to "transfer" (redirect) a
channel (Reported by Matt Jordan)
* ASTERISK-24363 - [patch] Add ability for Channel Drivers to
provide Presence State information (Reported by Gareth Palmer)
* ASTERISK-24554 - AMI/ARI: Generate events on connected line
changes (Reported by Matt Jordan)
* ASTERISK-24276 - [Patch] Option to make app MOH override channel
musicclass (Reported by Kristian Høgh)
* ASTERISK-23871 - RLS Tests: Implement RLS off-nominal tests
(Reported by Mark Michelson)
* ASTERISK-23823 - [patch] Option to keep queuerules in realtime
(Reported by Michael K.)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26227 - sqlalchemy error due to long identifier name
(Reported by Mark Michelson)
* ASTERISK-26221 - chan_sip: iLBC does not include correct mode
(Reported by Aaron Meriwether)
* ASTERISK-26216 - res_fax: Deadlock when detect fax while channel
executing Playback (Reported by Richard Mudgett)
* ASTERISK-26214 - Allow arbitrary time for fax detection to end
on a channel (Reported by Richard Mudgett)
* ASTERISK-23013 - [patch] Deadlock between 'sip show channels'
command and attended transfer handling (Reported by Ben
Smithurst)
* ASTERISK-26212 - [patch] Makefile: Retain XML Declaration and
DTD in docs. (Reported by Alexander Traud)
* ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in
conditional code. (Reported by Corey Farrell)
* ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence
number even on lost packets. (Reported by Alexander Traud)
* ASTERISK-26038 - 'make install' doesn't seem to install OS/X
init files (Reported by Tzafrir Cohen)
* ASTERISK-26133 - app_queue: Queue members receive multiple calls
(Reported by Richard Miller)
* ASTERISK-26196 - pbx: Time based includes can leak timezone
string (Reported by Corey Farrell)
* ASTERISK-26193 - chan_sip: reference leak in mwi_event_cb
(Reported by Corey Farrell)
* ASTERISK-26191 - threadpool: Leak on duplicate taskprocessor for
ast_threadpool_serializer_group (Reported by Corey Farrell)
* ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing
DTLS failure occurred on RTP instance (Reported by Edwin
Vandamme)
* ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf.
(Reported by Alexander Traud)
* ASTERISK-26160 - pjsip: Updated->Reachable during qualify
(Reported by Matt Jordan)
* ASTERISK-26177 - func_odbc: Database handle is kept when it
should be released (Reported by Leandro Dardini)
* ASTERISK-25289 - Build System does not respect CFLAGS and
CXXFLAGS when building menuselect (Reported by Jeffrey Walton)
* ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out
of bounds and bugs (Reported by Alexei Gradinari)
* ASTERISK-26184 - chan_sip: Reference leaks in error paths.
(Reported by Corey Farrell)
* ASTERISK-26181 - REF_DEBUG: Node object incorrectly logged
during duplicate replacement (Reported by Corey Farrell)
* ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported
by Joshua Colp)
* ASTERISK-26180 - PJSIP: provide valid tcp nodelay option for
reuse (Reported by Scott Griepentrog)
* ASTERISK-25772 - res_pjsip: Unexpected two BYE when answered
(Reported by Dmitriy Serov)
* ASTERISK-26099 - res_pjsip_pubsub: Crash when sending request
due to server timeout (Reported by Ross Beer)
* ASTERISK-26144 - Crash on loading codecs g729/g723 (Reported by
Alexei Gradinari)
* ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x
(Reported by George Joseph)
* ASTERISK-26021 - Build codecs siren7 and siren14 for Asterisk 13
(Reported by Daniel Denson)
* ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to
v21_details (Reported by Corey Farrell)
* ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a
self-comparison (Reported by George Joseph)
* ASTERISK-26138 - chan_unistim: Under FreeBSD, chan_unistim
generates a compile error (Reported by George Joseph)
* ASTERISK-26128 - Alembic scripts are failing (Reported by Mark
Michelson)
* ASTERISK-26139 - test_res_pjsip_scheduler: Compile failure if
pjproject isn't installed in a system location (Reported by
George Joseph)
* ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version.
(Reported by Alexander Traud)
* ASTERISK-26132 - PJSIP: provide transport type with received
messages (Reported by Scott Griepentrog)
* ASTERISK-26127 - res_pjsip_session: Crash due to race condition
between res_pjsip_session unload and timer (Reported by Joshua
Colp)
* ASTERISK-26045 - [patch]app_voicemail: fix bugs, imap mm_status
log change to debug (Reported by Alexei Gradinari)
* ASTERISK-26083 - ARI: Announcer channels staying around after
playback to a bridge is finished (Reported by Per Jensen)
* ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in
http.conf (Reported by Alexander Traud)
* ASTERISK-26097 - [patch] CLI: show maximum file descriptors
(Reported by Alexander Traud)
* ASTERISK-25262 - Memory leak when a caller channel does multiple
dials and CEL is enabled (Reported by Etienne Lessard)
* ASTERISK-26047 - ARI allows certain commands to run on down
channels. (Reported by Mark Michelson)
* ASTERISK-25959 -
http_media_cache/retrieve_cache_control_directives: Sporadic
failure (Reported by Joshua Colp)
* ASTERISK-26103 - cdr: Assert on 'dial end' event during a blond
transfer (Reported by George Joseph)
* ASTERISK-26092 - [Segfault] in res_rtp_asterisk.c:4268 after
Remotely bridged channels (Reported by Niklas Larsson)
* ASTERISK-26089 - Invalid security events during boot using PJSIP
Realtime (Reported by Scott Griepentrog)
* ASTERISK-26096 - res_hep: Crash when configuration file is
missing (Reported by Niklas Larsson)
* ASTERISK-26074 - res_odbc: Deadlock within UnixODBC (Reported by
Ross Beer)
* ASTERISK-26054 - Asterisk crashes (core dump) (Reported by B.
Davis)
* ASTERISK-26069 - Asterisk truncates To: header, dropping the
closing '>' (Reported by Vasil Kolev)
* ASTERISK-24436 - Missing header in res/res_srtp.c when compiling
against libsrtp-1.5.0 (Reported by Patrick Laimbock)
* ASTERISK-26091 - [patch] ar cru creates warning, instead use ar
cr (Reported by Alexander Traud)
* ASTERISK-26070 - ari/channels: Creating a local channel without
an originator adds all audio formats to it's capabilities
(Reported by George Joseph)
* ASTERISK-26078 - core: Memory leak in logging (Reported by
Etienne Lessard)
* ASTERISK-26065 - chan_pjsip: MWI NOTIFY contents not ordered
properly (Reported by Ross Beer)
* ASTERISK-26063 - ${PJSIP_HEADER(read,Call-ID)} does not work -
documentation needs clarification for when read/write is
possible (Reported by Private Name)
* ASTERISK-25777 - data race in threadpool (Reported by Badalian
Vyacheslav)
* ASTERISK-26053 - res_pjsip_outbound_publish: Crash when shutting
down (Reported by Joshua Colp)
* ASTERISK-26049 - res_pjsip: Crash when our own request timer
fires (Reported by Joshua Colp)
* ASTERISK-25669 - [patch]CURL incorrect trim for non ASCII
characters (Reported by Jesper)
* ASTERISK-26029 - parking: ast_parking_park_call should return
parking_space instead of parking_exten (Reported by Diederik de
Groot)
* ASTERISK-25938 - res_odbc: MySQL/MariaDB statement
LAST_INSERT_ID() always returns zero. (Reported by Edwin
Vandamme)
* ASTERISK-25941 - chan_pjsip: Crash on an immediate SIP final
response (Reported by Javier Riveros )
* ASTERISK-26014 - res_sorcery_astdb: Make tolerant of unknown
fields (Reported by Joshua Colp)
* ASTERISK-24986 - keepalive INFO packages ignored by asterisk
(Reported by Ilya Trikoz)
* ASTERISK-26034 - T.38 passthrough problem behind firewall due to
early nosignal packet (Reported by George Joseph)
* ASTERISK-26030 - call cut because of double Session-Expires
header in re-invite after proxy authentication is required
(Reported by George Joseph)
* ASTERISK-25964 - Outbound registrations created via ARI/push
configuration do not clean up outbound registrations currently
in flight (Reported by Matt Jordan)
* ASTERISK-26005 - res_pjsip: Multiple SIP messages are combined
into 1 TCP packet (Reported by Ross Beer)
* ASTERISK-25352 - res_hep_rtcp correlation_id is different then
res_hep (Reported by Kevin Scott Adams)
* ASTERISK-26007 - res_pjsip: Endpoints deleting early after
upgrade from 13.8.2 to 13.9 (Reported by Greg Siemon)
* ASTERISK-25990 - PJSIP TLS registration should respect
client_uri scheme when generating Contact URI (Reported by
Sebastian Damm)
* ASTERISK-26008 - app_followme does not delete recorded name
prompt (Reported by Tzafrir Cohen)
* ASTERISK-25978 - res_pjsip_authenticator_digest: Should not use
source port in nonce verification (Reported by Mark Michelson)
* ASTERISK-26004 - res_pjsip: The transport/method parameter is
ignored (Reported by George Joseph)
* ASTERISK-25999 - res_pjsip_dialog_info_body_generator: Remove
subscription requirement (Reported by Joshua Colp)
* ASTERISK-25993 - pjproject: Allow bundling to not require
everything it does (Reported by Joshua Colp)
* ASTERISK-25998 - file: Crash when using nativeformats (Reported
by Joshua Colp)
* ASTERISK-25826 - PJSIP / Sorcery slow load from realtime
(Reported by Ross Beer)
* ASTERISK-25956 - Compilation error in conditionally compiled
code in config_options.c (Reported by Chris Trobridge)
* ASTERISK-25968 - pjproject_bundled: Configure and make need to
be re-tested (Reported by George Joseph)
* ASTERISK-24463 - Voicemail email address corrupt or not sent
when message is in the process of being recorded during reload
(Reported by John Campbell)
* ASTERISK-25922 - res_pjsip_exten_state: Add configuration
support for publishing (Reported by Joshua Colp)
* ASTERISK-25970 - Segfault in pjsip_url_compare (Reported by
Dmitriy Serov)
* ASTERISK-25963 - func_odbc requires reconnect checks for stale
connections (Reported by Ross Beer)
* ASTERISK-25961 - tests/channels/SIP/sip_tls_call: Sporadic crash
when running test (Reported by Joshua Colp)
* ASTERISK-16115 - [patch] problem with ringinuse=no, queue
members receive sometimes two calls (Reported by nik600)
* ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir
only works if you manually add secret.conf yourself (Reported by
Jonathan R. Rose)
* ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions
are case sensitive to QueueName (Reported by Javier Acosta)
* ASTERISK-25951 - res_agi: run_agi eats frames it shouldn't
(Reported by George Joseph)
* ASTERISK-25950 - [patch]SIP channel does not send PeerStatus
events for autocreated peers (Reported by Kirill Katsnelson)
* ASTERISK-25927 - Removed option "registertrying" is still
documented in sip.conf.sample (Reported by Etienne Lessard)
* ASTERISK-25947 - Protocol transfers to stasis applications are
missing the StasisStart with the replace_channel object.
(Reported by Richard Mudgett)
* ASTERISK-24649 - Pushing of channel into bridge fails; Stasis
fails to get app name (Reported by John Bigelow)
* ASTERISK-24782 - StasisEnd event not present for channel that
was swapped out for another after completing attended transfer
(Reported by John Bigelow)
* ASTERISK-25942 - res_pjsip_caller_id: Transfer results in mixed
ConnectedLine information (Reported by George Joseph)
* ASTERISK-25928 - res_pjsip: URI validation done outside of PJSIP
thread (Reported by Joshua Colp)
* ASTERISK-25929 - res_pjsip_registrar: AOR_CONTACT_ADDED events
not raised (Reported by Joshua Colp)
* ASTERISK-25934 - chan_sip should not require sipregs or
updateable sippeers table unless rt (Reported by Jaco Kroon)
* ASTERISK-25888 - Frequent segfaults in function can_ring_entry()
of app_queue.c (Reported by Sébastien Couture)
* ASTERISK-25796 - res_pjsip: DOS/Crash when TCP/TLS sockets
exceed pjproject PJ_IOQUEUE_MAX_HANDLES (Reported by George
Joseph)
* ASTERISK-25707 - Long contact URIs or hostnames can crash
pjproject/Asterisk under certain conditions (Reported by George
Joseph)
* ASTERISK-25123 - Bracketed IPv6 Contact header parameter
unparsable with Asterisk/PJSIP (Reported by Anthony Messina)
* ASTERISK-25874 - app_voicemail: Stack buffer overflow in
test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
* ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE
without adding them to the local hangupcauses via
ast_channel_hangupcause_hash_set (Reported by Jaco Kroon)
* ASTERISK-25885 - res_pjsip: Race condition between adding
contact and automatic expiration (Reported by Joshua Colp)
* ASTERISK-25910 - pjproject: Via headers are not parsed when
"received" contains an IPv6 address (Reported by George Joseph)
* ASTERISK-25890 - Asterisk 13.8.0 alembic database update fails
(Reported by Harley Peters)
* ASTERISK-25894 - [patch] webrtc video broken due to missing
marker bits in RTP streams (Reported by Jacek Konieczny)
* ASTERISK-25881 - pbx: Add support for autohints (Reported by
Joshua Colp)
* ASTERISK-25854 - No audio after HOLD/RESUME - incorrect
a=recvonly in SDP from Asterisk (Reported by Robert McGilvray)
* ASTERISK-25868 - Sorcery "append to category" should allow
filters (Reported by Nick Repin)
* ASTERISK-25873 - res_pjsip: Bundled pjproject: compile error,
cannot find -lasteriskpj (Reported by Hans van Eijsden)
* ASTERISK-25882 - ARI: Crash can occur due to race condition when
attempting to operate on a hung up channel (Part 2) (Reported by
Richard Mudgett)
* ASTERISK-25867 - [patch] Video delay on app_echo (Reported by
Jacek Konieczny)
* ASTERISK-24605 - res_parking option parkeddynamic does not work
with the core Features 'parkcall' (DTMF initiated parking)
(Reported by Philip Correia)
* ASTERISK-24596 - Unclear how to use Park application with
res_parking 'parkeddynamic' enabled. Documentation? (Reported by
Philip Correia)
* ASTERISK-25825 - Crashes during shutdown when running CLI
commands (Reported by Mark Michelson)
* ASTERISK-24543 - Asterisk 13 responds to SIP Invite with all
possible codecs configured for peer as opposed to intersection
of configured codecs and offered codecs (Reported by Taylor
Hawkes)
* ASTERISK-25407 - Asterisk fails to log to multiple syslog
destinations (Reported by Elazar Broad)
* ASTERISK-25510 - [patch]Log to syslog failing (Reported by
Michael Newton)
* ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
data corruption (Reported by Gianluca Merlo)
* ASTERISK-25849 - chan_pjsip: transfers with direct media
sometimes drops audio (Reported by Kevin Harwell)
* ASTERISK-25814 - Segfault at f ip in res_pjsip_refer.so
(Reported by Sergio Medina Toledo)
* ASTERISK-25023 - Deadlock in chan_sip in
update_provisional_keepalive (Reported by Arnd Schmitter)
* ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
channel (Reported by Filip Frank)
* ASTERISK-25829 - res_pjsip: PJSIP does not accept spaces when
separating multiple AORs (Reported by Mateusz Kowalski)
* ASTERISK-25771 - ARI:Crash - Attended transfers of channels into
Stasis application. (Reported by Javier Riveros )
* ASTERISK-25830 - Revision 2451d4e breaks NAT (Reported by Sean
Bright)
* ASTERISK-25582 - Testsuite: Reactor timeout error in
tests/fax/pjsip/directmedia_reinvite_t38 (Reported by Matt
Jordan)
* ASTERISK-25811 - Unable to delete object from sorcery cache
(Reported by Ross Beer)
* ASTERISK-25800 - [patch] Calculate talktime when is first call
answered (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25727 - RPM build requires OPTIONAL_API cflag due to
PJSIP requirement (Reported by Gergely Dömsödi)
* ASTERISK-25337 - Crash on PJSIP_HEADER Add P-Asserted-Identity
when calling from Gosub (Reported by Jacques Peacock)
* ASTERISK-25738 - res_pjsip_pubsub: Crash while executing
OutboundSubscriptionDetail ami action (Reported by Kevin
Harwell)
* ASTERISK-25721 - [patch] res_phoneprov: memory leak and
heap-use-after-free (Reported by Badalian Vyacheslav)
* ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
returns garbage (Reported by Etienne Lessard)
* ASTERISK-25751 - res_pjsip: Support
pjsip_dlg_create_uas_and_inc_lock (Reported by Joshua Colp)
* ASTERISK-25606 - Core dump when using transports in sorcery
(Reported by Martin MouÄka)
* ASTERISK-20987 - non-admin users, who join muted conference are
not being muted (Reported by hristo)
* ASTERISK-25737 - res_pjsip_outbound_registration: line option
not in Alembic (Reported by Joshua Colp)
* ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
Vulnerability - Investigate vulnerability of HTTP server
(Reported by Alex A. Welzl)
* ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
udptl_rx_packet cause ast_frdup crash (Reported by Walter
Doekes)
* ASTERISK-25742 - Secondary IFP Packets can result in accessing
uninitialized pointers and a crash (Reported by Torrey Searle)
* ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
non-default timert1 (Reported by Alexander Traud)
* ASTERISK-25702 - PjSip realtime DB and Cache Errors since
upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported by
Nic Colledge)
* ASTERISK-25730 - build: make uninstall after make distclean
tries to remove root (Reported by George Joseph)
* ASTERISK-25725 - core: Incorrect XML documentation may result in
weird behavior (Reported by Joshua Colp)
* ASTERISK-25722 - ASAN & testsute: stack-buffer-overflow in
sip_sipredirect (Reported by Badalian Vyacheslav)
* ASTERISK-25709 - ARI: Crash can occur due to race condition when
attempting to operate on a hung up channel (Reported by Mark
Michelson)
* ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
by Badalian Vyacheslav)
* ASTERISK-25685 - infrastructure: Run alembic in Jenkins build
script (Reported by Joshua Colp)
* ASTERISK-25712 - Second call to already-on-call phone and
Asterisk sends "Ready" (Reported by Richard Mudgett)
* ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
(Reported by Badalian Vyacheslav)
* ASTERISK-25179 - CDR(billsec,f) and CDR(duration,f) report
incorrect values (Reported by Gianluca Merlo)
* ASTERISK-25611 - core: threadpool thread_timeout_thrash unit
test sporadically failing (Reported by Joshua Colp)
* ASTERISK-25686 - PJSIP: qualify_timeout is a double, database
schema is an integer (Reported by Marcelo Terres)
* ASTERISK-25700 - main/config: Clean config maps on shutdown.
(Reported by Corey Farrell)
* ASTERISK-25696 - bridge_basic: don't cache xferfailsound during
a transfer (Reported by Kevin Harwell)
* ASTERISK-25697 - bridge_basic: don't play an attended transfer
fail sound after target hangs up (Reported by Kevin Harwell)
* ASTERISK-25683 - res_ari: Asterisk fails to start if compiled
with MALLOC_DEBUG (Reported by yaron nahum)
* ASTERISK-24097 - Documentation - CHANNEL function help text
missing 'linkedid' argument (Reported by Steven T. Wheeler)
* ASTERISK-25690 - Hanging up when executing connected line sub
does not cause hangup (Reported by Joshua Colp)
* ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
reload' cause a crash (Reported by Sean Bright)
* ASTERISK-25632 - res_pjsip_sdp_rtp: RTP is sent from wrong IP
address when multihomed (Reported by Olivier Krief)
* ASTERISK-25637 - Multi homed server using wrong IP (Reported by
Daniel Journo)
* ASTERISK-25394 - pbx: Incorrect device and presence state when
changing hint details (Reported by Joshua Colp)
* ASTERISK-25640 - pbx: Deadlock on features reload and state
change hint. (Reported by Krzysztof Trempala)
* ASTERISK-25681 - devicestate: Engine thread is not shut down
(Reported by Corey Farrell)
* ASTERISK-25680 - manager: manager_channelvars is not cleaned at
shutdown (Reported by Corey Farrell)
* ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
Corey Farrell)
* ASTERISK-25675 - Endpoint not listed as Unreachable (Reported by
Daniel Journo)
* ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
by Corey Farrell)
* ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
Farrell)
* ASTERISK-25668 - res_pjsip: Deadlock in distributor (Reported by
Mark Michelson)
* ASTERISK-25664 - ast_format_cap_append_by_type leaks a reference
(Reported by Corey Farrell)
* ASTERISK-25647 - bug of cel_radius.c: wrong point of
ADD_VENDOR_CODE (Reported by Aaron An)
* ASTERISK-25137 - endpoint stasis messages are delivered twice
(Reported by Vitezslav Novy)
* ASTERISK-25116 - res_pjsip: Two PeerStatus AMI messages are
sent for every status change (Reported by George Joseph)
* ASTERISK-25641 - bridge: GOTO_ON_BLINDXFR doesn't work on
transfer initiated channel (Reported by Dmitry Melekhov)
* ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
Brandon)
* ASTERISK-25625 - res_sorcery_memory_cache: Add full backend
caching (Reported by Joshua Colp)
* ASTERISK-25601 - json: Audit reference usage and thread safety
(Reported by Joshua Colp)
* ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
sungtae kim)
* ASTERISK-25615 - res_pjsip: Setting transport async_operations >
1 causes segfault on tls transports (Reported by George Joseph)
* ASTERISK-25442 - using realtime (mysql) queue members are never
updated in wait_our_turn function (app_queue.c) (Reported by
Carlos Oliva)
* ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
thread of asterisk is not released (Reported by Hiroaki Komatsu)
* ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
Corey Farrell)
* ASTERISK-25619 - res_chan_stats not sending the correct
information to StatsD (Reported by Tyler Cambron)
* ASTERISK-24146 - [patch]No audio on WebRtc caller side when
answer waiting time is more than ~7sec (Reported by Aleksei
Kulakov)
* ASTERISK-25609 - [patch]Asterisk may crash when calling
ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
* ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
(Reported by Alexander Traud)
* ASTERISK-25616 - Warning with a Codec Module which supports PLC
with FEC (Reported by Alexander Traud)
* ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
Dudás József)
* ASTERISK-25608 - res_pjsip/contacts/statsd: Lifecycle events
aren't consistent (Reported by George Joseph)
* ASTERISK-25584 - [patch] format-attribute module: VP8 missing
(Reported by Alexander Traud)
* ASTERISK-25583 - [patch] format-attribute module: RFC 7587 (Opus
Codec) (Reported by Alexander Traud)
* ASTERISK-25498 - Asterisk crashes when negotiating g729 without
that module installed (Reported by Ben Langfeld)
* ASTERISK-25595 - Unescaped : in messge sent to statsd (Reported
by Niklas Larsson)
* ASTERISK-25598 - res_pjsip: Contact status messages are
printing a hash instead of the uri (Reported by George Joseph)
* ASTERISK-25600 - bridging: Inconsistency in BRIDGEPEER (Reported
by Jonathan Rose)
* ASTERISK-25476 - chan_sip loses registrations after a while
(Reported by Michael Keuter)
* ASTERISK-25593 - fastagi: record file closed after sending
result (Reported by Kevin Harwell)
* ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
it's assumed to (Reported by Walter Doekes)
* ASTERISK-25590 - CLI Usage info for 'pjsip send notify'
references incorrect config (Reported by Corey Farrell)
* ASTERISK-25165 - Testsuite - Sorcery memory cache leaks
(Reported by Corey Farrell)
* ASTERISK-25575 - res_pjsip: Dynamic outbound registrations
created via ARI are not loaded into memory on Asterisk
start/restart (Reported by Matt Jordan)
* ASTERISK-25545 - [patch] translation module gets cached not
joint format (Reported by Alexander Traud)
* ASTERISK-25573 - [patch] H.264 format attribute module: resets
whole SDP (Reported by Alexander Traud)
* ASTERISK-24958 - Forwarding loop detection inhibits certain
desirable scenarios (Reported by Mark Michelson)
* ASTERISK-25561 - app_queue.c line 6503 (try_calling): mutex
'qe->chan' freed more times than we've locked! (Reported by Alec
Davis)
* ASTERISK-25565 - DNS: System resolver only returns 1 record per
result (Reported by George Joseph)
* ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
Joshua Colp)
* ASTERISK-25160 - [patch] Opus Codec: SIP/SDP line fmtp missing
when called internally (Reported by Alexander Traud)
* ASTERISK-25535 - [patch] format creation on module load instead
of cache (Reported by Alexander Traud)
* ASTERISK-25449 - main/sched: Regression introduced by
5c713fdf18f causes erroneous duplicate RTCP messages; other
potential scheduling issues in chan_sip/chan_skinny (Reported by
Matt Jordan)
* ASTERISK-25546 - threadpool: Race condition between idle timeout
and activation (Reported by Joshua Colp)
* ASTERISK-25537 - [patch] format-attribute module: RFC or
internal defaults? (Reported by Alexander Traud)
* ASTERISK-25533 - [patch] buffer for ast_format_cap_get_names
only 64 bytes (Reported by Alexander Traud)
* ASTERISK-25373 - add documentation for CALLERID(pres) and also
the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
Doekes)
* ASTERISK-25528 - DNS: System resolver issues with TTL parse
(Reported by dtryba)
* ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
Walter Doekes)
* ASTERISK-24779 - Passthrough OPUS codec not working with
chan_pjsip (Reported by PowerPBX)
* ASTERISK-25522 - ARI: Crash when creating channel via ARI
originate with requesting channel (Reported by Matt Jordan)
* ASTERISK-25434 - Compiler flags not reported in 'core show
settings' despite usage during compilation (Reported by Rusty
Newton)
* ASTERISK-24106 - WebSockets Automatically decides what driver it
will use (Reported by Andrew Nagy)
* ASTERISK-25513 - Crash: malloc failed with high load of
subscriptions. (Reported by John Bigelow)
* ASTERISK-25505 - res_pjsip_pubsub: Crash on off-nominal when UAS
dialog can't be created (Reported by Joshua Colp)
* ASTERISK-25494 - build: GCC 5.1.x catches some new const, array
bounds and missing paren issues (Reported by George Joseph)
* ASTERISK-25485 - res_pjsip_outbound_registration: registration
stops due to 400 response (Reported by Kevin Harwell)
* ASTERISK-25486 - res_pjsip: Fix deadlock when validating URIs
(Reported by Joshua Colp)
* ASTERISK-7803 - [patch] Update the maximum packetization values
in frame.c (Reported by dea)
* ASTERISK-25484 - [patch] autoframing=yes has no effect (Reported
by Alexander Traud)
* ASTERISK-25308 - ari: Websocket leak (Reported by Joshua Colp)
* ASTERISK-25461 - Nested dialplan #includes don't work as
expected. (Reported by Richard Mudgett)
* ASTERISK-25455 - Deadlock of PJSIP realtime over
res_config_pgsql (Reported by mdu113)
* ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
(Reported by Olle Johansson)
* ASTERISK-25108 - configure check for older unbound library
(Reported by John Bigelow)
* ASTERISK-25435 - Asterisk periodically hangs. UDP Recv-Q greatly
exceeds zero. (Reported by Dmitriy Serov)
* ASTERISK-25451 - Broken video - erased rtp marker bit (Reported
by Stefan Engström)
* ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
exist in AstDB (Reported by Andrew Nagy)
* ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
parsing (Reported by ffs)
* ASTERISK-25404 - segfault/crash in chan_pjsip_hangup ... at
chan_pjsip.c (Reported by Chet Stevens)
* ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
(Reported by Bojan NemÄić)
* ASTERISK-25441 - Deadlock in res_sorcery_memory_cache. (Reported
by Richard Mudgett)
* ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
ICE is not enabled (Reported by Joshua Colp)
* ASTERISK-25383 - Core dumps on startup and shutdown with
MALLOC_DEBUG enabled (Reported by yaron nahum)
* ASTERISK-25423 - Caller gets no Connected line update during
call pickup. (Reported by Richard Mudgett)
* ASTERISK-25305 - Dynamic logger channels can be added multiple
times (Reported by Mark Michelson)
* ASTERISK-25418 - On-hold channels redirected out of a bridge
appear to still be on hold (Reported by Mark Michelson)
* ASTERISK-25384 - Regular Asterisk crashes when using Page
application. "user_data is NULL" (Reported by Chet Stevens)
* ASTERISK-25410 - app_record: RECORDED_FILE variable not being
populated (Reported by Kevin Harwell)
* ASTERISK-25396 - chan_sip: Extremely long callerid name causes
invalid SIP (Reported by Walter Doekes)
* ASTERISK-25399 - app_queue: AgentComplete event has wrong reason
(Reported by Kevin Harwell)
* ASTERISK-25185 - Segfault in app_queue on transfer scenarios
(Reported by Etienne Lessard)
* ASTERISK-25353 - [patch] Transcoding while different in Frame
size = Frames lost (Reported by Alexander Traud)
* ASTERISK-25325 - ARI PUT reload chan_sip HTTP response 404
(Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25390 - default_from_user can crash with certain
configuration backends (Reported by Mark Michelson)
* ASTERISK-25387 - res_pjsip_nat: Malformed REGISTER request
causes NAT'd Contact header to not be rewritten (Reported by
Matt Jordan)
* ASTERISK-25227 - No audio at in-band announcements in ooh323
channel (Reported by Alexandr Dranchuk)
* ASTERISK-25295 - res_pjsip crash - pjsip_uri_get_uri at
/usr/include/pjsip/sip_uri.h (Reported by Dmitriy Serov)
* ASTERISK-25381 - res_pjsip: AoRs deleted via ARI (or other
mechanism) do not destroy their related contacts (Reported by
Matt Jordan)
* ASTERISK-25369 - res_parking: ParkAndAnnounce - Inheritable
variables aren't applied to the announcer channel (Reported by
Jonathan Rose)
* ASTERISK-25367 - pbx: Long pattern match hints may cause "core
show hints" to crash (Reported by Joshua Colp)
* ASTERISK-25365 - Persistent subscriptions have extra
Content-Length/corrupted messages (Reported by Mark Michelson)
* ASTERISK-25356 - res_pjsip_sdp_rtp: Multiple keepalive scheduled
items may exist (Reported by Joshua Colp)
* ASTERISK-25355 - sched: ast_sched_del may return prematurely due
to spurious wakeup (Reported by Joshua Colp)
* ASTERISK-25318 -
tests/rest_api/applications/subscribe-endpoint/nominal/resource:
Sporadically failing (Reported by Joshua Colp)
* ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
cause on call pickup (Reported by Joshua Colp)
* ASTERISK-25342 - res_pjsip: Repeated usage of pj_gethostip may
block (Reported by Joshua Colp)
* ASTERISK-25341 - bridge: Hangups may get lost when executing
actions (Reported by Joshua Colp)
* ASTERISK-25339 - res_pjsip: Empty "auth" sections from
non-config backgrounds are interpreted as valid (Reported by
Matt Jordan)
* ASTERISK-25215 - Differences in queue.log between Set
QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
Gaetz)
* ASTERISK-25322 - Crash occurs when using MixMonitor with t() or
r() options. (Reported by Richard Mudgett)
* ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
for wrong or non existent peer on invite (Reported by Kevin
Harwell)
* ASTERISK-25312 - res_http_websocket: Terminate connection on
fatal cases (Reported by Joshua Colp)
* ASTERISK-25315 - DAHDI channels send shortened duration DTMF
tones. (Reported by Richard Mudgett)
* ASTERISK-25306 - Persistent subscriptions can save multiple SIP
messages at once, leading to potential crashes. (Reported by
Mark Michelson)
* ASTERISK-25309 - [patch] iLBC 20 advertised (Reported by
Alexander Traud)
* ASTERISK-25304 - res_pjsip: XML sanitization may write past
buffer (Reported by Joshua Colp)
* ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
Firefox 39 - add ECDH support and fallback to prime256v1
(Reported by Stefan Engström)
* ASTERISK-24988 - func_talkdetect: Test is bouncing sporadically
(Reported by Joshua Colp)
* ASTERISK-25181 - ARI: Channels added to Stasis application
during WebSocket creation don't receive a StasisStart event
(Reported by Matt Jordan)
* ASTERISK-25296 - RTP performance issue with several channel
drivers. (Reported by Richard Mudgett)
* ASTERISK-25297 - Crashes running
channels/pjsip/resolver/srv/failover/in_dialog testsuite tests
(Reported by Richard Mudgett)
* ASTERISK-25292 - Testuite:
tests/apps/bridge/bridge_wait/bridge_wait_e_options fails
(Reported by Kevin Harwell)
* ASTERISK-25271 - Parking & blind transfer: Transferer channel
not hung up if no MOH (Reported by Kevin Harwell)
* ASTERISK-25250 - chan_sip - Despite the channel being answered,
caller on a call established via Local channel continues to hear
ringback (Reported by Etienne Lessard)
* ASTERISK-25253 - confbridge volume options and other volume
controls such as func_volume don't work (Reported by Dmitriy
Serov)
* ASTERISK-25247 - choppy audio when spying on a g722 channel,
chan_sip or chan_pjsip (Reported by hristo)
* ASTERISK-25263 - [patch]cdr_adaptive_odbc: CDR insert failure
due to reversed if logic (Reported by Elazar Broad)
* ASTERISK-24867 - Docs for 'e' option in ResetCDR say to use
CDR_PROP instead, CDR_PROP docs are unclear (Reported by Rusty
Newton)
* ASTERISK-24853 - Documentation claims chan_sip outbound
registrations support WS or WSS as valid transports (not true)
(Reported by PSDK)
* ASTERISK-25242 - PJSIP: No audio when Asterisk inside NAT and
endpoints outside NAT - implement functionality similar to
chan_sip 'rtpkeepalive'? (Reported by Mark Michelson)
* ASTERISK-25258 - chan_pjsip: Incorrect format switch on received
RTP packet (Reported by Joshua Colp)
* ASTERISK-25257 - [patch]channels/sig_pri.h -> sig_pri_span ->
force_restart_unavailable_chans in wrong scope (Reported by
Patric Marschall)
* ASTERISK-24934 - [patch]Asterisk manager output does not escape
control characters (Reported by warren smith)
* ASTERISK-25255 - Missing AMI VarSet events when setting to an
empty string. (Reported by Richard Mudgett)
* ASTERISK-25254 - Crash if dialplan sets ATTENDEDTRANSFER to an
empty string before Park. (Reported by Richard Mudgett)
* ASTERISK-25183 - PJSIP: Crash on NULL channel in
chan_pjsip_incoming_response despite previous checks for NULL
channel (Reported by Matt Jordan)
* ASTERISK-25201 - Crash in PJSIP distributor on already free'd
threadpool (Reported by Matt Jordan)
* ASTERISK-25240 - bridge_native_rtp: Direct media wrongfully
started when completing attended transfer (Reported by Joshua
Colp)
* ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
(Reported by Rusty Newton)
* ASTERISK-25146 - DNS: Create system level resolver (Reported by
Joshua Colp)
* ASTERISK-22805 - res_rtp_asterisk: Crash when calling
BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
(Reported by Dmitry Burilov)
* ASTERISK-24550 - res_rtp_asterisk: Crash in
ast_rtp_on_ice_complete during DTLS handshake (Reported by
Osaulenko Alexander)
* ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
Badalian Vyacheslav)
* ASTERISK-24832 - [patch]DTLS-crashes within openssl (Reported
by Stefan Engström)
* ASTERISK-25127 - DTLS crashes following "Unable to cancel
schedule ID" in dtls_srtp_check_pending (Reported by Dade
Brandon)
* ASTERISK-25168 - Random Core Dumps on Asterisk 13.4 PJSIP, in
ast_channel_name at channel_internal_api.c (Reported by Carl
Fortin)
* ASTERISK-25076 - res_pjsip: Failover does not occur on
connection-less transport or 503 response (Reported by Joshua
Colp)
* ASTERISK-25226 - chan_sip: Channel leak in branch 13 on early
replaces call pickup (Reported by Walter Doekes)
* ASTERISK-25222 - Crash in recurring cancel callback called from
ast_dns_resolve_cancel on junk pointer (Reported by Matt Jordan)
* ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
(Reported by Walter Doekes)
* ASTERISK-25219 - [patch]Source and destination overlap in memcpy
in rtp_engine.c (Reported by Walter Doekes)
* ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
(Reported by Walter Doekes)
* ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
Bad file descriptor" (Reported by Barry Chern)
* ASTERISK-25202 - Hints extension state broken between 13.3.2 and
13.4 (Reported by cervajs)
* ASTERISK-25196 - res_pjsip_nat: rewrite_contact should not be
applied to Contact header when Record-Route headers are present
(Reported by Mark Michelson)
* ASTERISK-24907 - res_pjsip_outbound_registration: crash during
unload if registration attempts are still occuring (Reported by
Kevin Harwell)
* ASTERISK-25204 - res_pjsip_refer: Duplicated Referred-By or
Replaces headers on outbound INVITEs. (Reported by Mark
Michelson)
* ASTERISK-25189 - AMI: Add Linkedid header to standard channel
snapshot information. (Reported by Richard Mudgett)
* ASTERISK-25171 - Early completion of feature code attended
transfer results in intermittent one-way audio, "ghost ringing"
and robotic sound. (Reported by Rusty Newton)
* ASTERISK-25172 - Crash in channels/sip/sip blind
transfer/caller_refer_only test in
ast_format_cap_append_from_cap during ast_request (Reported by
Matt Jordan)
* ASTERISK-25180 - res_pjsip_mwi: Unsolicited MWI requires reload
(Reported by Joshua Colp)
* ASTERISK-25182 - [patch] on CLI sip reload, new codecs get
appended only (Reported by Alexander Traud)
* ASTERISK-25163 - Deadlock in chan_sip between reload of sip peer
container and MWI Stasis callback (Reported by Dmitriy Serov)
* ASTERISK-25091 - Asterisk REST API - bridge.addChannel crash
asterisk when calling channel hangup while adding to bridge
(Reported by Ilya Trikoz)
* ASTERISK-24900 - Manager event ParkedCallSwap is not documented
(Reported by Rusty Newton)
* ASTERISK-25162 - func_pjsip_aor: Leak of contact in iterator
(Reported by Corey Farrell)
* ASTERISK-25158 - res_pjsip: Add option to use AAL2 packing when
negotiating g.726 (Reported by Kevin Harwell)
* ASTERISK-24344 - CDR_PROP(disable) disables CDR only for first
dialed party (Reported by Janusz Karolak)
* ASTERISK-24443 - CDR fields (dst, dcontext) empty in transfer
call started from Macro (Reported by Arveno Santoro)
* ASTERISK-25154 - [patch]fromtag may need to be updated after
successful call dialog match (Reported by Damian Ivereigh)
* ASTERISK-25156 - chan_pjsip’s CHAN_START cel event lacks the
correct context and exten (Reported by cloos)
* ASTERISK-25157 - bridging: Performing a blonde transfer does not
result in connected line updates (Reported by Joshua Colp)
* ASTERISK-25087 - Asterisk segfault when using Directory
application with alias option and specific mailbox configuration
(Reported by Chet Stevens)
* ASTERISK-25115 - Crash related to func
sip_resolve_invoke_user_callback of res_pjsip/pjsip_resolver.c
(Reported by John Bigelow)
* ASTERISK-25096 - [patch]Segfault when registering over
websockets with PJSIP (in ast_sockaddr_isnull at
/include/asterisk/netsock2.h) (Reported by Josh Kitchens)
* ASTERISK-24963 - ASAN: heap-use-after-free with PJSIP and WSS
(Reported by Badalian Vyacheslav)
* ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
but asterisk doesn't detect it. (Reported by ibercom)
* ASTERISK-25094 - PBX core: Investigate thread safety issues
(Reported by Corey Farrell)
* ASTERISK-25113 - install_prereq in Debian 8 without "standard
system utilities" (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25148 - res_pjsip NULL channel audit (Reported by Mark
Michelson)
* ASTERISK-25131 - chan_pjsip: In-dialog authentication not
handled. (Reported by Richard Mudgett)
* ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
| adpcm | ipc10} (Reported by Badalian Vyacheslav)
* ASTERISK-25100 - asterisk coredump if host has an IPv6 address
that end with ::80 (Reported by Mark Petersen)
* ASTERISK-25122 - Large SIP packet received via pjsip over
websocket crashes Asterisk (Reported by Ivan Poddubny)
* ASTERISK-25121 - Stasis: Fix unsafe use of stasis_unsubscribe in
modules. (Reported by Corey Farrell)
* ASTERISK-25120 - Astobj2: Weakproxy subscriptions should be run
in reverse order. (Reported by Corey Farrell)
* ASTERISK-25105 - res_pjsip: Possible incompatibility between
qualify_timeout and pjproject-2.4 (Reported by George Joseph)
* ASTERISK-25117 - res_mwi_external_ami: Fix manager action
registrations. (Reported by Corey Farrell)
* ASTERISK-25112 - Logger: Configuration settings are not reset to
default during reload. (Reported by Corey Farrell)
* ASTERISK-24983 - IAX deadlock between hangup and scheduled
actions (ex. largrq) (Reported by Y Ateya)
* ASTERISK-24944 - main/audiohook.c change prevents G722 call
recording (Reported by Ronald Raikes)
* ASTERISK-25110 - res_resolver_unbound.c compilation failure:
SIGURG is undeclared in func unbound_resolver_stop (Reported by
John Bigelow)
* ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
or more digits (Reported by Makoto Dei)
* ASTERISK-25086 - [patch]PJSIP crashes if endpoint missing in
Dial() (Reported by snuffy)
* ASTERISK-25089 - res_pjsip_config_wizard: Variable specified in
templates aren't being processed correctly (Reported by George
Joseph)
* ASTERISK-25090 - CLI core show channel truncates cdr variables
(Reported by snuffy)
* ASTERISK-25083 - Message.c: Message channel becomes saturated
with frames leading to spammy log messages (Reported by Jonathan
Rose)
* ASTERISK-25085 - [patch]Potential crash after unload of
func_periodic_hook or test_message (Reported by Corey Farrell)
* ASTERISK-25082 - Asterisk deletes message after doing a playback
of an INBOX message using ast_vm_play when the Old folder is
full for that mailbox. (Reported by Jonathan Rose)
* ASTERISK-21893 - Segfault after call hangup, in
ast_channel_hangupcause_set, at channel_internal_api.c (Reported
by Aleksandr Gordeev)
* ASTERISK-25042 - asterisk.conf options override command-line
options. (Reported by Corey Farrell)
* ASTERISK-25074 - Regression: Recent clang-related change broke
cross compiling of Asterisk (Reported by Sebastian Kemper)
* ASTERISK-24442 - Outgoing call files don't work properly when
set in the future (Reported by tootai)
* ASTERISK-18252 - queue_log mysql time column data format
(Reported by Gareth Blades)
* ASTERISK-25041 - [patch]Broken column type checking in
res_config_mysql addon (Reported by Alexandre Fournier)
* ASTERISK-25057 - res_pjsip_pubsub: Crash in send_notify due to
invalid root pointer in sub_tree (Reported by Matt Jordan)
* ASTERISK-24938 - ARI Snoop Channel results in excessive
escalating CPU usage (Reported by George Ladoff)
* ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
ignore ISDN RESTART requests. (Reported by Richard Mudgett)
* ASTERISK-25003 - Asterisk crashes on attended transfer (using
feature) (Reported by Artem Volodin)
* ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
contain waiting time (Reported by Etienne Lessard)
* ASTERISK-25027 - Build System: Many ARI modules are missing
dependencies. (Reported by Corey Farrell)
* ASTERISK-25061 - pbx_config: Register manager actions with
module version of macro. (Reported by Corey Farrell)
* ASTERISK-24967 - Problem support schema for pgsql on CEL
(Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25025 - Periodic crashes (in
ast_channel_snapshot_create at stasis_channels.c) with Certified
Asterisk 13. (Reported by Chet Stevens)
* ASTERISK-25053 - Unit test category /main/presence missing
trailing slash. (Reported by Corey Farrell)
* ASTERISK-22708 - res_odbc.conf negative_connection_cache option
not respected, failover between DSNs doesn't work (Reported by
JoshE)
* ASTERISK-25054 - Formats interface's cannot be unregistered,
needs to hold modules until shutdown. (Reported by Corey
Farrell)
* ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
(Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25033 - Asterisk 13 (branch head) won't compile without
PJSip (Reported by Peter Whisker)
* ASTERISK-24896 - [patch] Using force black background leads to
colours not being reset (Reported by dant)
* ASTERISK-25048 - Astobj2: Initialization order wrong when both
refdebug and AO2_DEBUG are both enabled. (Reported by Corey
Farrell)
* ASTERISK-19608 - Asterisk-1.8.x starts rejecting calls with
cause code 44 after some time. (Reported by Denis Alberto
Martinez)
* ASTERISK-25037 - res_pjsip_outbound_registration: Potential
crash in off-nominal failure case when sending message (Reported
by Joshua Colp)
* ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
(Reported by Steve Davies)
* ASTERISK-22790 - check_modem_rate() may return incorrect rate
for V.27 (Reported by not here)
* ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
to minrate=2400, then res_fax refuse to load (Reported by David
Brillert)
* ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
which is disallowed in res_fax's check_modem_rate (Reported by
Matt Jordan)
* ASTERISK-25020 - Mismatched response to outgoing REGISTER
request (Reported by Mark Michelson)
* ASTERISK-25028 - Build System: Unneeded defines in
asterisk/buildopts.h (Reported by Corey Farrell)
* ASTERISK-25026 - Git conversion: Non-C files not switched to
ASTERISK_REGISTER_FILE (Reported by Corey Farrell)
* ASTERISK-24996 - chan_pjsip: Creating Channel Causes Asterisk to
Crash When Duplicate AOR Sections Exist in pjsip.conf (Reported
by Ashley Sanders)
* ASTERISK-25018 - pjsip show endpoints crashes asterisk when
qualified aors present (Reported by Ivan Poddubny)
* ASTERISK-24749 - ConfBridge: Wrong language on playing
conf-hasjoin and conf-hasleft when played to bridge (Reported by
Philippe Bolduc)
* ASTERISK-24845 - pjsip send notify not working with Cisco phone
(Reported by Carl Fortin)
* ASTERISK-25004 - Crash in authenticated reinvite after
originated T.38 FAX (Reported by Mark Michelson)
* ASTERISK-24999 - PJSIP crashes with malformed contact line
(Reported by snuffy)
* ASTERISK-24998 - res_corosync: res_corosync tries to load even
if res_corosync.conf is missing (Reported by George Joseph)
* ASTERISK-24997 - Astobj2: Some callers of __adjust_lock do not
pre-check the object (Reported by Corey Farrell)
* ASTERISK-24994 - dns: Query set unit tests are failing due to
race condition (Reported by Joshua Colp)
* ASTERISK-24982 - res_pjsip_mwi: Unsolicited MWI NOTIFY only sent
on mailbox changes (Reported by Joshua Colp)
* ASTERISK-24991 - Check for ao2_alloc failure in
__ast_channel_internal_alloc (Reported by Corey Farrell)
* ASTERISK-24895 - After hangup on the side of the ISDN network no
HangupRequest event comes for the dahdi channel. (Reported by
Andrew Zherdin)
* ASTERISK-24977 - Contacts that don't use qualify are being
marked as unavailable (Reported by George Joseph)
* ASTERISK-24774 - Segfault in ast_context_destroy with
extensions.ael and extensions.conf (Reported by Corey Farrell)
* ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when
channels have multiple native formats (Reported by Matt Jordan)
* ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
to Fail (Reported by Ashley Sanders)
* ASTERISK-24863 - res_pjsip: No endpoint events raised via AMI
when contacts cannot be reached/qualified (Reported by Dmitriy
Serov)
* ASTERISK-24869 - Asterisk segfaults on DAHDI attended transfer
due to application (appl) being NULL on unbridged channel
(Reported by viniciusfontes)
* ASTERISK-24970 - Crash in res_pjsip_pubsub handling of failed
notify (Reported by Scott Griepentrog)
* ASTERISK-13271 - menuselect sets defaults too late (Reported by
John Nemeth)
* ASTERISK-24959 - [patch]CLI command cdr show pgsql status
(Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-20524 - AMI improperly handles lines of exactly 1025
characters (Reported by David M. Lee)
* ASTERISK-24936 - New Feature: AO2 weakproxy objects (Reported by
Corey Farrell)
* ASTERISK-24954 - Git migration: Asterisk version numbers are
incompatible with the Test Suite (Reported by Matt Jordan)
* ASTERISK-17608 - func_aes.so cannot be loaded if res_crypto /
openssl not compiled (Reported by Warren Selby)
* ASTERISK-24928 - [patch]t38_udptl_maxdatagram in pjsip.conf not
honored (Reported by Juergen Spies)
* ASTERISK-24835 - Early Media Not working with Chan SIP and
Asterisk 13 (Reported by Andrew Nagy)
* ASTERISK-21777 - Asterisk tries to transcode video instead of
audio (Reported by Nick Ruggles)
* ASTERISK-24380 - core: Native formats are set to h264 with
certain audio/video codec configuration, resulting in path
translation WARNINGs (Reported by Matt Jordan)
* ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
into account (Reported by Frederic Van Espen)
* ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
short (Reported by Y Ateya)
* ASTERISK-24935 - res_pjsip_phoneprov_provider: Fix leaked
OBJ_MULTIPLE iterator. (Reported by Corey Farrell)
* ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
(Reported by Vadim)
* ASTERISK-24933 - T38 fails negotiation (Reported by Jonathan
Rose)
* ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
byte prefix bug (Reported by Matt Jordan)
* ASTERISK-21211 - chan_iax2 - unprotected access of
iaxs[peer->callno] potentially results in segfault (Reported by
Jaco Kroon)
* ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
(Reported by Christoph Timm)
* ASTERISK-24910 - "timer=no" and "timer=required" settings in
pjsip.conf fail (Reported by Ray Crumrine)
* ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
(Reported by Jeffrey C. Ollie)
* ASTERISK-24914 - Division by zero in file.c when playback of
voicemail with video as h264 (Reported by Marcello Ceschia)
* ASTERISK-24899 - Parking fall-through behavior different in 13
(Reported by Malcolm Davenport)
* ASTERISK-24937 - [patch]res_pjsip_messaging: Messages may be
sent out of order (Reported by Mark Michelson)
* ASTERISK-24920 - Asterisk handles duplicate SIP requests as if
they were each a new request (Reported by Mark Michelson)
* ASTERISK-24781 - PJSIP: Unnecessary 180 Ringing messages sent
with undesireabe consequences. (Reported by Richard Mudgett)
* ASTERISK-24857 - [patch] "timing test", pjsip incoming/outgoing
calls, voicemail prompts and recordings all fail when using the
kqueue timer source on FreeBSD 10.x (Reported by Justin T.
Gibbs)
* ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
detection in ast_malloc (Reported by Timo Teräs)
* ASTERISK-24142 - CCSS: crash during shutdown due to device
lookup in destroyed container (Reported by David Brillert)
* ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
core restart now (Reported by Peter Katzmann)
* ASTERISK-24805 - [patch] - ASAN: Race condition
(heap-use-after-free) on asterisk closing (Reported by Badalian
Vyacheslav)
* ASTERISK-24881 - ast_register_atexit should only be used when
absolutely needed (Reported by Corey Farrell)
* ASTERISK-24731 - res_pjsip_session cannot be unloaded (Reported
by Corey Farrell)
* ASTERISK-24864 - app_confbridge: file playback blocks dtmf
(Reported by Kevin Harwell)
* ASTERISK-14233 - [patch] Buddies are always auto-registered when
processing the roster (Reported by Simon Arlott)
* ASTERISK-24780 - [patch] - Buddies are always auto-registered
when processing the roster (Reported by Simon Arlott)
* ASTERISK-24879 - [patch]Compilation fails due to 64bit time
under OpenBSD (Reported by snuffy)
* ASTERISK-24880 - [patch]Compilation under OpenBSD (Reported by
snuffy)
* ASTERISK-21765 - [patch] - FILE function's length argument
counts from beginning of file rather than the offset (Reported
by John Zhong)
* ASTERISK-24817 - init_logger_chain: unreachable code block
(Reported by Corey Farrell)
* ASTERISK-24882 - chan_sip: Improve usage of REF_DEBUG (Reported
by Corey Farrell)
* ASTERISK-24876 - Investigate reference leaks from
tests/channels/local/local_optimize_away (Reported by Corey
Farrell)
* ASTERISK-24840 - res_pjsip: conflicting endpoint identifiers
(Reported by Kevin Harwell)
* ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
Atis Lezdins)
* ASTERISK-18708 - func_curl hangs channel under load (Reported by
Dave Cabot)
* ASTERISK-21038 - Bad command completion of "core set debug
channel" (Reported by Richard Kenner)
* ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
by Frank DiGennaro)
* ASTERISK-24872 - [patch] AMI PJSIPShowEndpoint closes AMI
connection on error (Reported by Dmitriy Serov)
* ASTERISK-23666 - CLONE - nested functions aren't portable
(Reported by Diederik de Groot)
* ASTERISK-20399 - Compilation on some systems requires the
-fnested-functions flag (Reported by David M. Lee)
* ASTERISK-20850 - [patch]Nested functions aren't portable.
Adapting RAII_VAR to use clang/llvm blocks to get the
same/similar functionality. (Reported by Diederik de Groot)
* ASTERISK-24807 - Missing mandatory field Max-Forwards (Reported
by Anatoli)
* ASTERISK-24808 - res_config_odbc: Improper escaping of
backslashes occurs with MySQL (Reported by Javier Acosta)
* ASTERISK-23390 - NewExten Event with application AGI shows up
before and after AGI runs (Reported by Benjamin Keith Ford)
* ASTERISK-24786 - [patch] - Asterisk terminates when playing a
voicemail stored in LDAP (Reported by Graham Barnett)
* ASTERISK-24739 - [patch] - Out of files -- call fails --
numerous files with inodes from under /usr/share/zoneinfo,
mostly posixrules (Reported by Ed Hynan)
* ASTERISK-24755 - Asterisk sends unexpected early BYE to
transferrer during attended transfer when using a Stasis bridge
(Reported by John Bigelow)
* ASTERISK-24830 - res_rtp_asterisk.c checks USE_PJPROJECT not
HAVE_PJPROJECT (Reported by Stefan Engström)
* ASTERISK-24825 - Caller ID not recognized using
Centrex/Distinctive dialing (Reported by Richard Mudgett)
* ASTERISK-17588 - Caller ID on TDM410P *UK* PSTN (Reported by
Daniel Flounders)
* ASTERISK-24838 - chan_sip: Locking inversion occurs when
building a peer causes a peer poke during request handling
(Reported by Richard Mudgett)
* ASTERISK-24751 - Integer values in json payload to ARI cause
asterisk to crash (Reported by jeffrey putnam)
* ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
* ASTERISK-18105 - most of asterisk modules are unbuildable in
cygwin environment (Reported by feyfre)
* ASTERISK-21845 - maxcalls exceeded, Asterisk sends out 480 and
also BYE (Reported by Tony Ching)
* ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
error response and BYE are sent to the caller (Reported by
Makoto Dei)
* ASTERISK-23214 - chan_sip WARNING message 'We are requesting
SRTP for audio, but they responded without it' is ambiguous and
wrong in some cases (Reported by Rusty Newton)
* ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
fail (Reported by Terry Wilson)
* ASTERISK-20233 - SRTP not working with some devices (Eg
Grandstream gxv3175) - Message "Can't provide secure audio
requested in SDP offer" (Reported by tootai)
* ASTERISK-22748 - SRTP Crypto Offer With Lifetime Not Accepted
(Reported by Alejandro Mejia)
* ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
thread ID being passed to pthread_kill (Reported by JoshE)
* ASTERISK-24812 - ARI: Creating channels through /channels
resource always uses SLIN, which results in unneeded transcoding
(Reported by Matt Jordan)
* ASTERISK-24797 - bridge_softmix: G.729 codec license held
(Reported by Kevin Harwell)
* ASTERISK-24677 - ARI GET variable on channel provides unhelpful
response on non-existent variable (Reported by Joshua Colp)
* ASTERISK-24785 - 'Expires' header missing from 200 OK on
REGISTER (Reported by Ross Beer)
* ASTERISK-24499 - Need more explicit debug when PJSIP dialstring
is invalid (Reported by Rusty Newton)
* ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
(Reported by Ashley Sanders)
* ASTERISK-24796 - Codecs and bucket schema's prevent module
unload (Reported by Corey Farrell)
* ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
OSX with 64 bit integers (Reported by Corey Farrell)
* ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
for playing back messages stored in IMAP - play_message: No
origtime (Reported by Graham Barnett)
* ASTERISK-22670 - Asterisk crashes when processing ISDN AoC
Events (Reported by klaus3000)
* ASTERISK-24689 - Segfault on hangup after outgoing PRI-Euroisdn
call (Reported by Marcel Manz)
* ASTERISK-24740 - [patch]Segmentation fault on aoc-e event
(Reported by Panos Gkikakis)
* ASTERISK-24799 - [patch] make fails with undefined reference to
SSLv3_client_method (Reported by Alexander Traud)
* ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
(Reported by Corey Farrell)
* ASTERISK-24700 - CRASH: NULL channel is being passed to
ast_bridge_transfer_attended() (Reported by Zane Conkle)
* ASTERISK-24791 - Crash in ast_rtcp_write_report (Reported by
JoshE)
* ASTERISK-24085 - Documentation - We should remove or further
document the 'contact' section in pjsip.conf (Reported by Rusty
Newton)
* ASTERISK-24632 - install_prereq script installs pjproject
without IPv6 support (Reported by Rusty Newton)
* ASTERISK-24685 - "pjsip show version" CLI command (Reported by
Joshua Colp)
* ASTERISK-24768 - res_timing_pthread: file descriptor leak
(Reported by Matthias Urlichs)
* ASTERISK-24612 - res_pjsip: No information if a required sorcery
wizard is not loaded (Reported by Joshua Colp)
* ASTERISK-24716 - Improve pjsip log messages for presence
subscription failure (Reported by Rusty Newton)
* ASTERISK-24771 - ${CHANNEL(pjsip)} - segfault (Reported by
Niklas Larsson)
* ASTERISK-24727 - PJSIP: Crash experienced during multi-Asterisk
transfer scenario. (Reported by Mark Michelson)
* ASTERISK-24015 - app_transfer fails with PJSIP channels
(Reported by Private Name)
* ASTERISK-24741 - dtls_handler causes Asterisk to crash (Reported
by Zane Conkle)
* ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
fully disconnect underlying socket, leading to events being
dropped with no additional information (Reported by Matt Jordan)
* ASTERISK-24752 - Crash in bridge_manager_service_req when bridge
is destroyed by ARI during shutdown (Reported by Richard
Mudgett)
* ASTERISK-24772 - ODBC error in realtime sippeers when device
unregisters under MariaDB (Reported by Richard Miller)
* ASTERISK-24479 - Enable REF_DEBUG for module references
(Reported by Corey Farrell)
* ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
res_odbc (Reported by ibercom)
* ASTERISK-24769 - res_pjsip_sdp_rtp: Local ICE candidates leaked
(Reported by Matt Jordan)
* ASTERISK-24748 - res_pjsip: If wizards explicitly configured in
sorcery.conf false ERROR messages may occur (Reported by Joshua
Colp)
* ASTERISK-24616 - Crash in res_format_attr_h264 due to invalid
string copy (Reported by Yura Kocyuba)
* ASTERISK-24737 - When agent not logged in, agent status shows
unavailable, queue status shows agent invalid (Reported by
Richard Mudgett)
* ASTERISK-24635 - PJSIP outbound PUBLISH crashes when no response
is ever received (Reported by Marco Paland)
* ASTERISK-24736 - Memory Leak Fixes (Reported by Mark Michelson)
* ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
Stephan Eisvogel)
* ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
versions (Reported by Jared Biel)
* ASTERISK-24666 - Security Vulnerability: RTP not closed after
sip call using unsupported codec (Reported by Y Ateya)
* ASTERISK-24676 - Security Vulnerability: URL request injection
in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
* ASTERISK-24729 - Outbound registration not occuring on new
registrations after reload. (Reported by Richard Mudgett)
* ASTERISK-24728 - tcptls: Bad file descriptor error when
reloading chan_sip (Reported by Kevin Harwell)
* ASTERISK-24721 - manager: ModuleLoad action incorrectly reports
'module not found' during a Reload operation (Reported by Matt
Jordan)
* ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
by Kevin Harwell)
* ASTERISK-24485 - res_pjsip cannot be unloaded or shutdown
(Reported by Corey Farrell)
* ASTERISK-24719 - ConfBridge recording channels get stuck when
recording started/stopped more than once (Reported by Richard
Mudgett)
* ASTERISK-24723 - confbridge: CLI command 'confbridge list XXXX'
no longer displays user menus (Reported by Matt Jordan)
* ASTERISK-24539 - Compile fails on OSX because of sem_timedwait
in bridge_channel.c (Reported by George Joseph)
* ASTERISK-24544 - Compile fails on OSX Yosemite because of
incorrect detection of htonll and ntohll (Reported by George
Joseph)
* ASTERISK-24231 - crash: CLI execution of realtime destroy
sippeers id 1 causes crash due to NULL name provided to
ast_variable (Reported by Niklas Larsson)
* ASTERISK-24626 - Voicemail passwords not being stored in ARA
(Reported by Paddy Grice)
* ASTERISK-24693 - Investigate and fix memory leaks in Asterisk
(Reported by Kevin Harwell)
* 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 Engström)
* 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 Høgh)
* 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 Engström)
* 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 Cisárik)
* 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 Lainé)
* 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 Lainé)
* 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 <a class="moz-txt-link-rfc2396E" href="mailto:anonymous@anonymous.invalid"><anonymous@anonymous.invalid></a> (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 Høgh)
* 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 Lainé)
* 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:
<a class="moz-txt-link-freetext" href="http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.0.0-beta1">http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.0.0-beta1</a>
Thank you for your continued support of Asterisk!
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<pre class="moz-signature" cols="72">--
Borja SIXTO (co-founder/manager)
Mail: <a class="moz-txt-link-abbreviated" href="mailto:borja.sixto@ulex.fr">borja.sixto@ulex.fr</a>
Tel: +33(0)4 28 35 03 44 Ext. 11
Ulex Innovative Systems | 4 rue des Prairies, 38460 St Romain de Jalionas, France
<a class="moz-txt-link-abbreviated" href="http://www.ulex.fr">www.ulex.fr</a></pre>
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