[asterisk-dev] Asterisk 14.0.0-beta1 Now Available
Borja SIXTO
borja.sixto at ulex.fr
Mon Aug 1 16:07:04 CDT 2016
Hi,
The configure not tests the libsrtp dependencies.
If you not install it you will see the error :
sdp_srtp.c:37:72: fatal error: srtp/srtp.h: No such file or directory
#include <srtp/srtp.h> /* for SRTP_MAX_KEY_LEN, etc */
For debian use :
# apt-get install libsrtp0-dev
Regards,
Borja SIXTO from Ulex.fr
Le 27/07/2016 à 21:52, Asterisk Development Team a écrit :
> The Asterisk Development Team has announced the first beta of
> Asterisk 14.0.0. This beta is available for immediate
> download at http://downloads.asterisk.org/pub/telephony/asterisk
>
> The release of Asterisk 14.0.0-beta1 resolves several issues reported by the
> community and would have not been possible without your participation.
> Thank you!
>
> The following are the issues resolved in this beta:
>
> New Features made in this release:
> -----------------------------------
> * ASTERISK-25904 - PJSIP: add contact.updated event (Reported by
> Alexei Gradinari)
> * ASTERISK-26058 - [Patch] Add uptime and last reloaded to
> FullyBooted AMI event (Reported by Niklas Larsson)
> * ASTERISK-25925 - Allow Early Bridges on ARI Dials (Reported by
> Mark Michelson)
> * ASTERISK-26068 - Multicast RTP Options (Reported by Mark
> Michelson)
> * ASTERISK-26042 - ARI: Allow downloading of the media associated
> with a stored recording (Reported by Matt Jordan)
> * ASTERISK-25425 - logger: Add JSON structured logging (Reported
> by Matt Jordan)
> * ASTERISK-25900 - PJSIP Endpoint IP Access Controls (Reported by
> Alexei Gradinari)
> * ASTERISK-25972 - res_pjsip_exten_state: Use body generator to
> publish extension state (Reported by Richard Mudgett)
> * ASTERISK-25889 - ARI: Add separate "create" and "dial"
> operations for channels (Reported by Mark Michelson)
> * ASTERISK-25803 - [patch] chan_sip: Optionally supply
> fromuser/fromdomain in SIP dial string (Reported by Walter
> Doekes)
> * ASTERISK-24919 - res_pjsip_config_wizard: Ability to write
> contents to file (Reported by Ray Crumrine)
> * ASTERISK-25670 - Add regcontext to PJSIP (Reported by Daniel
> Journo)
> * ASTERISK-25660 - Add sipp-sendfax.xml and spandspflow2pcap.py to
> contrib/scripts. (Reported by Walter Doekes)
> * ASTERISK-25591 - [patch] Complete List of Header Files
> (#include): iwyu (Reported by Alexander Traud)
> * ASTERISK-25551 - [patch]Ability to add channel to an existing
> bridge by specifying an existing channel prefix (Reported by
> Alec Davis)
> * ASTERISK-25419 - Dialplan Application for Integration of StatsD
> (Reported by Ashley Sanders)
> * ASTERISK-25549 - Confbridge: Add participant timeout option
> (Reported by Mark Michelson)
> * ASTERISK-24922 - ARI: Add the ability to intercept hold and
> raise an event (Reported by Matt Jordan)
> * ASTERISK-25479 - Allow CDR's to be modified before being
> dispatched to engines (Reported by Jonh Wendell)
> * ASTERISK-25480 - [patch]Add field PauseReason on
> QueueMemberStatus (Reported by Rodrigo Ramirez Norambuena)
> * ASTERISK-25377 - res_pjsip: Change default "From user" from UUID
> to something more palatable (Reported by Mark Michelson)
> * ASTERISK-25252 - ARI: Add the ability to manipulate log channels
> (Reported by Matt Jordan)
> * ASTERISK-25259 - chan_pjsip: Add rtptimeout support (Reported by
> Joshua Colp)
> * ASTERISK-25238 - ARI: Support push configuration (Reported by
> Matt Jordan)
> * ASTERISK-25173 - ARI: Add the ability to load/reload/unload an
> Asterisk module (Reported by Matt Jordan)
> * ASTERISK-25006 - [patch] Add support set character for quoted
> identifiers (Reported by Rodrigo Ramirez Norambuena)
> * ASTERISK-23186 - [patch] Add usegmtime option to cel_pgsql
> (Reported by Rodrigo Ramirez Norambuena)
> * ASTERISK-24931 - dns: Add support for SRV records. (Reported by
> Joshua Colp)
> * ASTERISK-24834 - DNS Overhaul: Implement the proposed core API -
> sync/async functions, resolver registration (Reported by Matt
> Jordan)
> * ASTERISK-24836 - DNS Overhaul: Write a Resolver Implementation
> (Reported by Matt Jordan)
> * ASTERISK-22591 - [patch]Prevent Asterisk from writing received
> SMS content in log (Reported by Jan Juergens)
> * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
> (Reported by Dwayne Hubbard)
> * ASTERISK-24703 - ARI: Add the ability to "transfer" (redirect) a
> channel (Reported by Matt Jordan)
> * ASTERISK-24363 - [patch] Add ability for Channel Drivers to
> provide Presence State information (Reported by Gareth Palmer)
> * ASTERISK-24554 - AMI/ARI: Generate events on connected line
> changes (Reported by Matt Jordan)
> * ASTERISK-24276 - [Patch] Option to make app MOH override channel
> musicclass (Reported by Kristian Høgh)
> * ASTERISK-23871 - RLS Tests: Implement RLS off-nominal tests
> (Reported by Mark Michelson)
> * ASTERISK-23823 - [patch] Option to keep queuerules in realtime
> (Reported by Michael K.)
>
> Bugs fixed in this release:
> -----------------------------------
> * ASTERISK-26227 - sqlalchemy error due to long identifier name
> (Reported by Mark Michelson)
> * ASTERISK-26221 - chan_sip: iLBC does not include correct mode
> (Reported by Aaron Meriwether)
> * ASTERISK-26216 - res_fax: Deadlock when detect fax while channel
> executing Playback (Reported by Richard Mudgett)
> * ASTERISK-26214 - Allow arbitrary time for fax detection to end
> on a channel (Reported by Richard Mudgett)
> * ASTERISK-23013 - [patch] Deadlock between 'sip show channels'
> command and attended transfer handling (Reported by Ben
> Smithurst)
> * ASTERISK-26212 - [patch] Makefile: Retain XML Declaration and
> DTD in docs. (Reported by Alexander Traud)
> * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in
> conditional code. (Reported by Corey Farrell)
> * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence
> number even on lost packets. (Reported by Alexander Traud)
> * ASTERISK-26038 - 'make install' doesn't seem to install OS/X
> init files (Reported by Tzafrir Cohen)
> * ASTERISK-26133 - app_queue: Queue members receive multiple calls
> (Reported by Richard Miller)
> * ASTERISK-26196 - pbx: Time based includes can leak timezone
> string (Reported by Corey Farrell)
> * ASTERISK-26193 - chan_sip: reference leak in mwi_event_cb
> (Reported by Corey Farrell)
> * ASTERISK-26191 - threadpool: Leak on duplicate taskprocessor for
> ast_threadpool_serializer_group (Reported by Corey Farrell)
> * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing
> DTLS failure occurred on RTP instance (Reported by Edwin
> Vandamme)
> * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf.
> (Reported by Alexander Traud)
> * ASTERISK-26160 - pjsip: Updated->Reachable during qualify
> (Reported by Matt Jordan)
> * ASTERISK-26177 - func_odbc: Database handle is kept when it
> should be released (Reported by Leandro Dardini)
> * ASTERISK-25289 - Build System does not respect CFLAGS and
> CXXFLAGS when building menuselect (Reported by Jeffrey Walton)
> * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out
> of bounds and bugs (Reported by Alexei Gradinari)
> * ASTERISK-26184 - chan_sip: Reference leaks in error paths.
> (Reported by Corey Farrell)
> * ASTERISK-26181 - REF_DEBUG: Node object incorrectly logged
> during duplicate replacement (Reported by Corey Farrell)
> * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported
> by Joshua Colp)
> * ASTERISK-26180 - PJSIP: provide valid tcp nodelay option for
> reuse (Reported by Scott Griepentrog)
> * ASTERISK-25772 - res_pjsip: Unexpected two BYE when answered
> (Reported by Dmitriy Serov)
> * ASTERISK-26099 - res_pjsip_pubsub: Crash when sending request
> due to server timeout (Reported by Ross Beer)
> * ASTERISK-26144 - Crash on loading codecs g729/g723 (Reported by
> Alexei Gradinari)
> * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x
> (Reported by George Joseph)
> * ASTERISK-26021 - Build codecs siren7 and siren14 for Asterisk 13
> (Reported by Daniel Denson)
> * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to
> v21_details (Reported by Corey Farrell)
> * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a
> self-comparison (Reported by George Joseph)
> * ASTERISK-26138 - chan_unistim: Under FreeBSD, chan_unistim
> generates a compile error (Reported by George Joseph)
> * ASTERISK-26128 - Alembic scripts are failing (Reported by Mark
> Michelson)
> * ASTERISK-26139 - test_res_pjsip_scheduler: Compile failure if
> pjproject isn't installed in a system location (Reported by
> George Joseph)
> * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version.
> (Reported by Alexander Traud)
> * ASTERISK-26132 - PJSIP: provide transport type with received
> messages (Reported by Scott Griepentrog)
> * ASTERISK-26127 - res_pjsip_session: Crash due to race condition
> between res_pjsip_session unload and timer (Reported by Joshua
> Colp)
> * ASTERISK-26045 - [patch]app_voicemail: fix bugs, imap mm_status
> log change to debug (Reported by Alexei Gradinari)
> * ASTERISK-26083 - ARI: Announcer channels staying around after
> playback to a bridge is finished (Reported by Per Jensen)
> * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in
> http.conf (Reported by Alexander Traud)
> * ASTERISK-26097 - [patch] CLI: show maximum file descriptors
> (Reported by Alexander Traud)
> * ASTERISK-25262 - Memory leak when a caller channel does multiple
> dials and CEL is enabled (Reported by Etienne Lessard)
> * ASTERISK-26047 - ARI allows certain commands to run on down
> channels. (Reported by Mark Michelson)
> * ASTERISK-25959 -
> http_media_cache/retrieve_cache_control_directives: Sporadic
> failure (Reported by Joshua Colp)
> * ASTERISK-26103 - cdr: Assert on 'dial end' event during a blond
> transfer (Reported by George Joseph)
> * ASTERISK-26092 - [Segfault] in res_rtp_asterisk.c:4268 after
> Remotely bridged channels (Reported by Niklas Larsson)
> * ASTERISK-26089 - Invalid security events during boot using PJSIP
> Realtime (Reported by Scott Griepentrog)
> * ASTERISK-26096 - res_hep: Crash when configuration file is
> missing (Reported by Niklas Larsson)
> * ASTERISK-26074 - res_odbc: Deadlock within UnixODBC (Reported by
> Ross Beer)
> * ASTERISK-26054 - Asterisk crashes (core dump) (Reported by B.
> Davis)
> * ASTERISK-26069 - Asterisk truncates To: header, dropping the
> closing '>' (Reported by Vasil Kolev)
> * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling
> against libsrtp-1.5.0 (Reported by Patrick Laimbock)
> * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar
> cr (Reported by Alexander Traud)
> * ASTERISK-26070 - ari/channels: Creating a local channel without
> an originator adds all audio formats to it's capabilities
> (Reported by George Joseph)
> * ASTERISK-26078 - core: Memory leak in logging (Reported by
> Etienne Lessard)
> * ASTERISK-26065 - chan_pjsip: MWI NOTIFY contents not ordered
> properly (Reported by Ross Beer)
> * ASTERISK-26063 - ${PJSIP_HEADER(read,Call-ID)} does not work -
> documentation needs clarification for when read/write is
> possible (Reported by Private Name)
> * ASTERISK-25777 - data race in threadpool (Reported by Badalian
> Vyacheslav)
> * ASTERISK-26053 - res_pjsip_outbound_publish: Crash when shutting
> down (Reported by Joshua Colp)
> * ASTERISK-26049 - res_pjsip: Crash when our own request timer
> fires (Reported by Joshua Colp)
> * ASTERISK-25669 - [patch]CURL incorrect trim for non ASCII
> characters (Reported by Jesper)
> * ASTERISK-26029 - parking: ast_parking_park_call should return
> parking_space instead of parking_exten (Reported by Diederik de
> Groot)
> * ASTERISK-25938 - res_odbc: MySQL/MariaDB statement
> LAST_INSERT_ID() always returns zero. (Reported by Edwin
> Vandamme)
> * ASTERISK-25941 - chan_pjsip: Crash on an immediate SIP final
> response (Reported by Javier Riveros )
> * ASTERISK-26014 - res_sorcery_astdb: Make tolerant of unknown
> fields (Reported by Joshua Colp)
> * ASTERISK-24986 - keepalive INFO packages ignored by asterisk
> (Reported by Ilya Trikoz)
> * ASTERISK-26034 - T.38 passthrough problem behind firewall due to
> early nosignal packet (Reported by George Joseph)
> * ASTERISK-26030 - call cut because of double Session-Expires
> header in re-invite after proxy authentication is required
> (Reported by George Joseph)
> * ASTERISK-25964 - Outbound registrations created via ARI/push
> configuration do not clean up outbound registrations currently
> in flight (Reported by Matt Jordan)
> * ASTERISK-26005 - res_pjsip: Multiple SIP messages are combined
> into 1 TCP packet (Reported by Ross Beer)
> * ASTERISK-25352 - res_hep_rtcp correlation_id is different then
> res_hep (Reported by Kevin Scott Adams)
> * ASTERISK-26007 - res_pjsip: Endpoints deleting early after
> upgrade from 13.8.2 to 13.9 (Reported by Greg Siemon)
> * ASTERISK-25990 - PJSIP TLS registration should respect
> client_uri scheme when generating Contact URI (Reported by
> Sebastian Damm)
> * ASTERISK-26008 - app_followme does not delete recorded name
> prompt (Reported by Tzafrir Cohen)
> * ASTERISK-25978 - res_pjsip_authenticator_digest: Should not use
> source port in nonce verification (Reported by Mark Michelson)
> * ASTERISK-26004 - res_pjsip: The transport/method parameter is
> ignored (Reported by George Joseph)
> * ASTERISK-25999 - res_pjsip_dialog_info_body_generator: Remove
> subscription requirement (Reported by Joshua Colp)
> * ASTERISK-25993 - pjproject: Allow bundling to not require
> everything it does (Reported by Joshua Colp)
> * ASTERISK-25998 - file: Crash when using nativeformats (Reported
> by Joshua Colp)
> * ASTERISK-25826 - PJSIP / Sorcery slow load from realtime
> (Reported by Ross Beer)
> * ASTERISK-25956 - Compilation error in conditionally compiled
> code in config_options.c (Reported by Chris Trobridge)
> * ASTERISK-25968 - pjproject_bundled: Configure and make need to
> be re-tested (Reported by George Joseph)
> * ASTERISK-24463 - Voicemail email address corrupt or not sent
> when message is in the process of being recorded during reload
> (Reported by John Campbell)
> * ASTERISK-25922 - res_pjsip_exten_state: Add configuration
> support for publishing (Reported by Joshua Colp)
> * ASTERISK-25970 - Segfault in pjsip_url_compare (Reported by
> Dmitriy Serov)
> * ASTERISK-25963 - func_odbc requires reconnect checks for stale
> connections (Reported by Ross Beer)
> * ASTERISK-25961 - tests/channels/SIP/sip_tls_call: Sporadic crash
> when running test (Reported by Joshua Colp)
> * ASTERISK-16115 - [patch] problem with ringinuse=no, queue
> members receive sometimes two calls (Reported by nik600)
> * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir
> only works if you manually add secret.conf yourself (Reported by
> Jonathan R. Rose)
> * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions
> are case sensitive to QueueName (Reported by Javier Acosta)
> * ASTERISK-25951 - res_agi: run_agi eats frames it shouldn't
> (Reported by George Joseph)
> * ASTERISK-25950 - [patch]SIP channel does not send PeerStatus
> events for autocreated peers (Reported by Kirill Katsnelson)
> * ASTERISK-25927 - Removed option "registertrying" is still
> documented in sip.conf.sample (Reported by Etienne Lessard)
> * ASTERISK-25947 - Protocol transfers to stasis applications are
> missing the StasisStart with the replace_channel object.
> (Reported by Richard Mudgett)
> * ASTERISK-24649 - Pushing of channel into bridge fails; Stasis
> fails to get app name (Reported by John Bigelow)
> * ASTERISK-24782 - StasisEnd event not present for channel that
> was swapped out for another after completing attended transfer
> (Reported by John Bigelow)
> * ASTERISK-25942 - res_pjsip_caller_id: Transfer results in mixed
> ConnectedLine information (Reported by George Joseph)
> * ASTERISK-25928 - res_pjsip: URI validation done outside of PJSIP
> thread (Reported by Joshua Colp)
> * ASTERISK-25929 - res_pjsip_registrar: AOR_CONTACT_ADDED events
> not raised (Reported by Joshua Colp)
> * ASTERISK-25934 - chan_sip should not require sipregs or
> updateable sippeers table unless rt (Reported by Jaco Kroon)
> * ASTERISK-25888 - Frequent segfaults in function can_ring_entry()
> of app_queue.c (Reported by Sébastien Couture)
> * ASTERISK-25796 - res_pjsip: DOS/Crash when TCP/TLS sockets
> exceed pjproject PJ_IOQUEUE_MAX_HANDLES (Reported by George
> Joseph)
> * ASTERISK-25707 - Long contact URIs or hostnames can crash
> pjproject/Asterisk under certain conditions (Reported by George
> Joseph)
> * ASTERISK-25123 - Bracketed IPv6 Contact header parameter
> unparsable with Asterisk/PJSIP (Reported by Anthony Messina)
> * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
> test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
> * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE
> without adding them to the local hangupcauses via
> ast_channel_hangupcause_hash_set (Reported by Jaco Kroon)
> * ASTERISK-25885 - res_pjsip: Race condition between adding
> contact and automatic expiration (Reported by Joshua Colp)
> * ASTERISK-25910 - pjproject: Via headers are not parsed when
> "received" contains an IPv6 address (Reported by George Joseph)
> * ASTERISK-25890 - Asterisk 13.8.0 alembic database update fails
> (Reported by Harley Peters)
> * ASTERISK-25894 - [patch] webrtc video broken due to missing
> marker bits in RTP streams (Reported by Jacek Konieczny)
> * ASTERISK-25881 - pbx: Add support for autohints (Reported by
> Joshua Colp)
> * ASTERISK-25854 - No audio after HOLD/RESUME - incorrect
> a=recvonly in SDP from Asterisk (Reported by Robert McGilvray)
> * ASTERISK-25868 - Sorcery "append to category" should allow
> filters (Reported by Nick Repin)
> * ASTERISK-25873 - res_pjsip: Bundled pjproject: compile error,
> cannot find -lasteriskpj (Reported by Hans van Eijsden)
> * ASTERISK-25882 - ARI: Crash can occur due to race condition when
> attempting to operate on a hung up channel (Part 2) (Reported by
> Richard Mudgett)
> * ASTERISK-25867 - [patch] Video delay on app_echo (Reported by
> Jacek Konieczny)
> * ASTERISK-24605 - res_parking option parkeddynamic does not work
> with the core Features 'parkcall' (DTMF initiated parking)
> (Reported by Philip Correia)
> * ASTERISK-24596 - Unclear how to use Park application with
> res_parking 'parkeddynamic' enabled. Documentation? (Reported by
> Philip Correia)
> * ASTERISK-25825 - Crashes during shutdown when running CLI
> commands (Reported by Mark Michelson)
> * ASTERISK-24543 - Asterisk 13 responds to SIP Invite with all
> possible codecs configured for peer as opposed to intersection
> of configured codecs and offered codecs (Reported by Taylor
> Hawkes)
> * ASTERISK-25407 - Asterisk fails to log to multiple syslog
> destinations (Reported by Elazar Broad)
> * ASTERISK-25510 - [patch]Log to syslog failing (Reported by
> Michael Newton)
> * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
> data corruption (Reported by Gianluca Merlo)
> * ASTERISK-25849 - chan_pjsip: transfers with direct media
> sometimes drops audio (Reported by Kevin Harwell)
> * ASTERISK-25814 - Segfault at f ip in res_pjsip_refer.so
> (Reported by Sergio Medina Toledo)
> * ASTERISK-25023 - Deadlock in chan_sip in
> update_provisional_keepalive (Reported by Arnd Schmitter)
> * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
> channel (Reported by Filip Frank)
> * ASTERISK-25829 - res_pjsip: PJSIP does not accept spaces when
> separating multiple AORs (Reported by Mateusz Kowalski)
> * ASTERISK-25771 - ARI:Crash - Attended transfers of channels into
> Stasis application. (Reported by Javier Riveros )
> * ASTERISK-25830 - Revision 2451d4e breaks NAT (Reported by Sean
> Bright)
> * ASTERISK-25582 - Testsuite: Reactor timeout error in
> tests/fax/pjsip/directmedia_reinvite_t38 (Reported by Matt
> Jordan)
> * ASTERISK-25811 - Unable to delete object from sorcery cache
> (Reported by Ross Beer)
> * ASTERISK-25800 - [patch] Calculate talktime when is first call
> answered (Reported by Rodrigo Ramirez Norambuena)
> * ASTERISK-25727 - RPM build requires OPTIONAL_API cflag due to
> PJSIP requirement (Reported by Gergely Dömsödi)
> * ASTERISK-25337 - Crash on PJSIP_HEADER Add P-Asserted-Identity
> when calling from Gosub (Reported by Jacques Peacock)
> * ASTERISK-25738 - res_pjsip_pubsub: Crash while executing
> OutboundSubscriptionDetail ami action (Reported by Kevin
> Harwell)
> * ASTERISK-25721 - [patch] res_phoneprov: memory leak and
> heap-use-after-free (Reported by Badalian Vyacheslav)
> * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
> returns garbage (Reported by Etienne Lessard)
> * ASTERISK-25751 - res_pjsip: Support
> pjsip_dlg_create_uas_and_inc_lock (Reported by Joshua Colp)
> * ASTERISK-25606 - Core dump when using transports in sorcery
> (Reported by Martin 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 <anonymous at anonymous.invalid> (Reported by Anthony
> Messina)
> * ASTERISK-25767 - [patch] Add check to configure for sanitizes
> (Reported by Badalian Vyacheslav)
> * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
> core set (Reported by Rusty Newton)
> * ASTERISK-25627 - Easily Preventable Compile Warning (Reported by
> Diederik de Groot)
> * ASTERISK-25558 - [patch]chan_sip option 'notifyringing' doc fix
> and addition of 'notifyringingprio' (Reported by Ward van
> Wanrooij)
> * ASTERISK-25618 - res_pjsip: Check for readability of TLS files
> at startup (Reported by George Joseph)
> * ASTERISK-25581 - [patch]Add value reason a pause on CLI
> (Reported by Rodrigo Ramirez Norambuena)
> * ASTERISK-25572 - Endpoints: Add StatsD stats for Asterisk
> endpoints (Reported by Matt Jordan)
> * ASTERISK-25571 - PJSIP: Add StatsD stats for some common PJSIP
> objects (Reported by Matt Jordan)
> * ASTERISK-25518 - taskprocessor: Add high water mark (Reported by
> Jonathan Rose)
> * ASTERISK-25495 - [patch] Prevent old-update packages on
> repository Debian systems (Reported by Rodrigo Ramirez
> Norambuena)
> * ASTERISK-25477 - pjsip show "command" like [criteria] (Reported
> by Bryant Zimmerman)
> * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
> configure (Reported by Badalian Vyacheslav)
> * ASTERISK-24870 - ARI: Subscriptions to bridges generally not
> super useful (Reported by Matt Jordan)
> * ASTERISK-25405 - [patch] CLI: core show fd: add timestamp
> (Reported by Alexander Traud)
> * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
> defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
> * ASTERISK-25256 - [patch]Post AMI VarSet to empty string events
> when Asterisk deletes a dialplan variable. (Reported by Richard
> Mudgett)
> * ASTERISK-25040 - pbx: Improve performance of reloads by making
> hint destruction more performant (Reported by Matt Jordan)
> * ASTERISK-25067 - Sorcery Caching: Implement a new caching module
> (Reported by Matt Jordan)
> * ASTERISK-25114 - res_pjsip: Add AMI events for chan_pjsip
> contact lifecycle changes (Reported by George Joseph)
> * ASTERISK-25072 - res_pjsip_outbound_registration: line
> functionality. Additional check for using the request URI
> (Reported by Dmitriy Serov)
> * ASTERISK-24815 - [patch] Enable TLS Dual-Certificates (ECC+RSA)
> (Reported by Alexander Traud)
> * ASTERISK-25063 - [patch]add X.509 subject alternative name
> support to Asterisk TLS support (Reported by Maciej Szmigiero)
> * ASTERISK-25044 - sorcery: Add ability to insert a new wizard
> into an object type's list (Reported by George Joseph)
> * ASTERISK-24892 - Super Awesome Company sound prompts (Reported
> by Rusty Newton)
> * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
> Hjelm)
> * ASTERISK-25049 - CLI: Enable automatic references to modules
> (Reported by Corey Farrell)
> * ASTERISK-25056 - Modules: Make ast_module_info->self available
> to auxiliary sources. (Reported by Corey Farrell)
> * ASTERISK-25045 - vector: Add new capabilities and unit tests
> (Reported by George Joseph)
> * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
> (Reported by Alexander Traud)
> * ASTERISK-24706 - [patch]add auto-dtmf mode for pjsip (Reported
> by yaron nahum)
> * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
> Diederik de Groot)
> * ASTERISK-25051 - Remove unneeded uses of optional_api providers.
> (Reported by Corey Farrell)
> * ASTERISK-24974 - Astobj2: Allow reference debugging to be
> enabled/disabled by config. (Reported by Corey Farrell)
> * ASTERISK-24980 - cdr_adaptive_odbc: refactor lines to
> concatenate of columns name (Reported by Rodrigo Ramirez
> Norambuena)
> * ASTERISK-24947 - res_pjsip: Add a PJSIP resolver using core DNS
> (Reported by Joshua Colp)
> * ASTERISK-24965 - cel_pgsql - log_error string references CDR
> instead of CEL (Reported by Rodrigo Ramirez Norambuena)
> * ASTERISK-24960 - Build System: Create MOD_ADD_SOURCE macro for
> module Makefiles (Reported by Corey Farrell)
> * ASTERISK-24939 - [patch]IAX make calltoken expiration time
> configurable (Reported by Y Ateya)
> * ASTERISK-24918 - pjsip: add CLI options to display global and
> system configuration (Reported by Scott Griepentrog)
> * ASTERISK-24862 - [patch] Support in-dialog OPTIONS (Reported by
> yaron nahum)
> * ASTERISK-24802 - stasis: set a channel variable on websocket
> disconnect error (Reported by Kevin Harwell)
> * ASTERISK-24133 - [patch]Please support Clang; Allow no-exec
> stacks (Reported by Jeffrey Walton)
> * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
> Couldn't find mailbox %s in context (Reported by Graham Barnett)
> * ASTERISK-24813 - asterisk.c: #if statement in listener()
> confuses code folding editors (Reported by Corey Farrell)
> * ASTERISK-24811 - asterisk-publication sorcery object does not
> use realtime (Reported by Matt Hoskins)
> * ASTERISK-24745 - [patch]Add no_answer to ARI hangup causes
> (Reported by Ben Merrills)
> * ASTERISK-24316 - For httpd server, need option to define server
> name for security purposes (Reported by Andrew Nagy)
> * ASTERISK-24671 - Missing docs for the CDR AMI Event (Reported by
> Dan Jenkins)
> * ASTERISK-24575 - [patch]Make capath work for res_pjsip (Reported
> by cloos)
> * ASTERISK-24678 - [PATCH] Added atxfer* settings to
> features.conf.sample (Reported by Niklas Larsson)
> * ASTERISK-24412 - [patch]Incomplete channel originate/continue
> handling with ARI (Reported by Nir Simionovich (GreenfieldTech -
> Israel))
> * ASTERISK-24351 - [patch] Allow passing options and command to
> MixMonitor when recording in ConfBridge (Reported by Gareth
> Palmer)
> * ASTERISK-24553 - ARI/AMI: Include language in standard channel
> snapshot output (Reported by Matt Jordan)
> * ASTERISK-24552 - ARI: Allow associating a channel as an
> initiator of an Origination for record keeping purposes
> (Reported by Matt Jordan)
> * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
> lookups (Reported by Birger "WIMPy" Harzenetter)
> * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
> recordings (Reported by Ben Smithurst)
> * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
> column in the cel_odbc module (Reported by Etienne Lessard)
> * ASTERISK-24128 - [Patch] Adding default dtls settings (Reported
> by Michael K.)
> * ASTERISK-24279 - Documentation: Clarify the behaviour of the CDR
> property 'unanswered' (Reported by Matt Jordan)
> * ASTERISK-23512 - Inaccurate comment in manager.conf.sample
> (Reported by Richard Miller)
> * ASTERISK-24365 - [Patch] Dialplan function to get first/head
> caller channel on queue (Reported by Kristian 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:
>
> http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.0.0-beta1
>
> Thank you for your continued support of Asterisk!
>
>
>
--
Borja SIXTO (co-founder/manager)
Mail: borja.sixto at ulex.fr
Tel: +33(0)4 28 35 03 44 Ext. 11
Ulex Innovative Systems | 4 rue des Prairies, 38460 St Romain de Jalionas, France
www.ulex.fr
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