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 byAlexei Gradinari)
* ASTERISK-26058 - [Patch] Add uptime and last reloaded toFullyBooted AMI event (Reported by Niklas Larsson)
* ASTERISK-25925 - Allow Early Bridges on ARI Dials (Reported byMark Michelson)
* ASTERISK-26068 - Multicast RTP Options (Reported by MarkMichelson)
* ASTERISK-26042 - ARI: Allow downloading of the media associatedwith a stored recording (Reported by Matt Jordan)
* ASTERISK-25425 - logger: Add JSON structured logging (Reportedby Matt Jordan)
* ASTERISK-25900 - PJSIP Endpoint IP Access Controls (Reported byAlexei Gradinari)
* ASTERISK-25972 - res_pjsip_exten_state: Use body generator topublish 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 supplyfromuser/fromdomain in SIP dial string (Reported by WalterDoekes)
* ASTERISK-24919 - res_pjsip_config_wizard: Ability to writecontents to file (Reported by Ray Crumrine)
* ASTERISK-25670 - Add regcontext to PJSIP (Reported by DanielJourno)
* ASTERISK-25660 - Add sipp-sendfax.xml and spandspflow2pcap.py tocontrib/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 existingbridge by specifying an existing channel prefix (Reported byAlec 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 andraise an event (Reported by Matt Jordan)
* ASTERISK-25479 - Allow CDR's to be modified before beingdispatched to engines (Reported by Jonh Wendell)
* ASTERISK-25480 - [patch]Add field PauseReason onQueueMemberStatus (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25377 - res_pjsip: Change default "From user" from UUIDto 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 byJoshua Colp)
* ASTERISK-25238 - ARI: Support push configuration (Reported byMatt Jordan)
* ASTERISK-25173 - ARI: Add the ability to load/reload/unload anAsterisk module (Reported by Matt Jordan)
* ASTERISK-25006 - [patch] Add support set character for quotedidentifiers (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 byJoshua Colp)
* ASTERISK-24834 - DNS Overhaul: Implement the proposed core API -sync/async functions, resolver registration (Reported by MattJordan)
* ASTERISK-24836 - DNS Overhaul: Write a Resolver Implementation(Reported by Matt Jordan)
* ASTERISK-22591 - [patch]Prevent Asterisk from writing receivedSMS 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) achannel (Reported by Matt Jordan)
* ASTERISK-24363 - [patch] Add ability for Channel Drivers toprovide Presence State information (Reported by Gareth Palmer)
* ASTERISK-24554 - AMI/ARI: Generate events on connected linechanges (Reported by Matt Jordan)
* ASTERISK-24276 - [Patch] Option to make app MOH override channelmusicclass (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 channelexecuting Playback (Reported by Richard Mudgett)
* ASTERISK-26214 - Allow arbitrary time for fax detection to endon a channel (Reported by Richard Mudgett)
* ASTERISK-23013 - [patch] Deadlock between 'sip show channels'command and attended transfer handling (Reported by BenSmithurst)
* ASTERISK-26212 - [patch] Makefile: Retain XML Declaration andDTD in docs. (Reported by Alexander Traud)
* ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used inconditional code. (Reported by Corey Farrell)
* ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequencenumber even on lost packets. (Reported by Alexander Traud)
* ASTERISK-26038 - 'make install' doesn't seem to install OS/Xinit 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 timezonestring (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 forast_threadpool_serializer_group (Reported by Corey Farrell)
* ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causingDTLS failure occurred on RTP instance (Reported by EdwinVandamme)
* 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 itshould be released (Reported by Leandro Dardini)
* ASTERISK-25289 - Build System does not respect CFLAGS andCXXFLAGS when building menuselect (Reported by Jeffrey Walton)
* ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, outof 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 loggedduring duplicate replacement (Reported by Corey Farrell)
* ASTERISK-26179 - chan_sip: Second T.38 request fails (Reportedby Joshua Colp)
* ASTERISK-26180 - PJSIP: provide valid tcp nodelay option forreuse (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 requestdue to server timeout (Reported by Ross Beer)
* ASTERISK-26144 - Crash on loading codecs g729/g723 (Reported byAlexei 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 tov21_details (Reported by Corey Farrell)
* ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught aself-comparison (Reported by George Joseph)
* ASTERISK-26138 - chan_unistim: Under FreeBSD, chan_unistimgenerates a compile error (Reported by George Joseph)
* ASTERISK-26128 - Alembic scripts are failing (Reported by MarkMichelson)
* ASTERISK-26139 - test_res_pjsip_scheduler: Compile failure ifpjproject isn't installed in a system location (Reported byGeorge Joseph)
* ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version.(Reported by Alexander Traud)
* ASTERISK-26132 - PJSIP: provide transport type with receivedmessages (Reported by Scott Griepentrog)
* ASTERISK-26127 - res_pjsip_session: Crash due to race conditionbetween res_pjsip_session unload and timer (Reported by JoshuaColp)
* ASTERISK-26045 - [patch]app_voicemail: fix bugs, imap mm_statuslog change to debug (Reported by Alexei Gradinari)
* ASTERISK-26083 - ARI: Announcer channels staying around afterplayback to a bridge is finished (Reported by Per Jensen)
* ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS inhttp.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 multipledials and CEL is enabled (Reported by Etienne Lessard)
* ASTERISK-26047 - ARI allows certain commands to run on downchannels. (Reported by Mark Michelson)
* ASTERISK-25959 -http_media_cache/retrieve_cache_control_directives: Sporadicfailure (Reported by Joshua Colp)
* ASTERISK-26103 - cdr: Assert on 'dial end' event during a blondtransfer (Reported by George Joseph)
* ASTERISK-26092 - [Segfault] in res_rtp_asterisk.c:4268 afterRemotely bridged channels (Reported by Niklas Larsson)
* ASTERISK-26089 - Invalid security events during boot using PJSIPRealtime (Reported by Scott Griepentrog)
* ASTERISK-26096 - res_hep: Crash when configuration file ismissing (Reported by Niklas Larsson)
* ASTERISK-26074 - res_odbc: Deadlock within UnixODBC (Reported byRoss Beer)
* ASTERISK-26054 - Asterisk crashes (core dump) (Reported by B.Davis)
* ASTERISK-26069 - Asterisk truncates To: header, dropping theclosing '>' (Reported by Vasil Kolev)
* ASTERISK-24436 - Missing header in res/res_srtp.c when compilingagainst libsrtp-1.5.0 (Reported by Patrick Laimbock)
* ASTERISK-26091 - [patch] ar cru creates warning, instead use arcr (Reported by Alexander Traud)
* ASTERISK-26070 - ari/channels: Creating a local channel withoutan originator adds all audio formats to it's capabilities(Reported by George Joseph)
* ASTERISK-26078 - core: Memory leak in logging (Reported byEtienne Lessard)
* ASTERISK-26065 - chan_pjsip: MWI NOTIFY contents not orderedproperly (Reported by Ross Beer)
* ASTERISK-26063 - ${PJSIP_HEADER(read,Call-ID)} does not work -documentation needs clarification for when read/write ispossible (Reported by Private Name)
* ASTERISK-25777 - data race in threadpool (Reported by BadalianVyacheslav)
* ASTERISK-26053 - res_pjsip_outbound_publish: Crash when shuttingdown (Reported by Joshua Colp)
* ASTERISK-26049 - res_pjsip: Crash when our own request timerfires (Reported by Joshua Colp)
* ASTERISK-25669 - [patch]CURL incorrect trim for non ASCIIcharacters (Reported by Jesper)
* ASTERISK-26029 - parking: ast_parking_park_call should returnparking_space instead of parking_exten (Reported by Diederik deGroot)
* ASTERISK-25938 - res_odbc: MySQL/MariaDB statementLAST_INSERT_ID() always returns zero. (Reported by EdwinVandamme)
* ASTERISK-25941 - chan_pjsip: Crash on an immediate SIP finalresponse (Reported by Javier Riveros )
* ASTERISK-26014 - res_sorcery_astdb: Make tolerant of unknownfields (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 toearly nosignal packet (Reported by George Joseph)
* ASTERISK-26030 - call cut because of double Session-Expiresheader in re-invite after proxy authentication is required(Reported by George Joseph)
* ASTERISK-25964 - Outbound registrations created via ARI/pushconfiguration do not clean up outbound registrations currentlyin flight (Reported by Matt Jordan)
* ASTERISK-26005 - res_pjsip: Multiple SIP messages are combinedinto 1 TCP packet (Reported by Ross Beer)
* ASTERISK-25352 - res_hep_rtcp correlation_id is different thenres_hep (Reported by Kevin Scott Adams)
* ASTERISK-26007 - res_pjsip: Endpoints deleting early afterupgrade from 13.8.2 to 13.9 (Reported by Greg Siemon)
* ASTERISK-25990 - PJSIP TLS registration should respectclient_uri scheme when generating Contact URI (Reported bySebastian Damm)
* ASTERISK-26008 - app_followme does not delete recorded nameprompt (Reported by Tzafrir Cohen)
* ASTERISK-25978 - res_pjsip_authenticator_digest: Should not usesource port in nonce verification (Reported by Mark Michelson)
* ASTERISK-26004 - res_pjsip: The transport/method parameter isignored (Reported by George Joseph)
* ASTERISK-25999 - res_pjsip_dialog_info_body_generator: Removesubscription requirement (Reported by Joshua Colp)
* ASTERISK-25993 - pjproject: Allow bundling to not requireeverything it does (Reported by Joshua Colp)
* ASTERISK-25998 - file: Crash when using nativeformats (Reportedby Joshua Colp)
* ASTERISK-25826 - PJSIP / Sorcery slow load from realtime(Reported by Ross Beer)
* ASTERISK-25956 - Compilation error in conditionally compiledcode in config_options.c (Reported by Chris Trobridge)
* ASTERISK-25968 - pjproject_bundled: Configure and make need tobe re-tested (Reported by George Joseph)
* ASTERISK-24463 - Voicemail email address corrupt or not sentwhen message is in the process of being recorded during reload(Reported by John Campbell)
* ASTERISK-25922 - res_pjsip_exten_state: Add configurationsupport for publishing (Reported by Joshua Colp)
* ASTERISK-25970 - Segfault in pjsip_url_compare (Reported byDmitriy Serov)
* ASTERISK-25963 - func_odbc requires reconnect checks for staleconnections (Reported by Ross Beer)
* ASTERISK-25961 - tests/channels/SIP/sip_tls_call: Sporadic crashwhen running test (Reported by Joshua Colp)
* ASTERISK-16115 - [patch] problem with ringinuse=no, queuemembers receive sometimes two calls (Reported by nik600)
* ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldironly works if you manually add secret.conf yourself (Reported byJonathan R. Rose)
* ASTERISK-25954 - Manager QueueSummary and QueueStatus Actionsare 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 PeerStatusevents for autocreated peers (Reported by Kirill Katsnelson)
* ASTERISK-25927 - Removed option "registertrying" is stilldocumented in sip.conf.sample (Reported by Etienne Lessard)
* ASTERISK-25947 - Protocol transfers to stasis applications aremissing the StasisStart with the replace_channel object.(Reported by Richard Mudgett)
* ASTERISK-24649 - Pushing of channel into bridge fails; Stasisfails to get app name (Reported by John Bigelow)
* ASTERISK-24782 - StasisEnd event not present for channel thatwas swapped out for another after completing attended transfer(Reported by John Bigelow)
* ASTERISK-25942 - res_pjsip_caller_id: Transfer results in mixedConnectedLine information (Reported by George Joseph)
* ASTERISK-25928 - res_pjsip: URI validation done outside of PJSIPthread (Reported by Joshua Colp)
* ASTERISK-25929 - res_pjsip_registrar: AOR_CONTACT_ADDED eventsnot raised (Reported by Joshua Colp)
* ASTERISK-25934 - chan_sip should not require sipregs orupdateable 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 socketsexceed pjproject PJ_IOQUEUE_MAX_HANDLES (Reported by GeorgeJoseph)
* ASTERISK-25707 - Long contact URIs or hostnames can crashpjproject/Asterisk under certain conditions (Reported by GeorgeJoseph)
* ASTERISK-25123 - Bracketed IPv6 Contact header parameterunparsable with Asterisk/PJSIP (Reported by Anthony Messina)
* ASTERISK-25874 - app_voicemail: Stack buffer overflow intest_voicemail_notify_endl (Reported by Badalian Vyacheslav)
* ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODEwithout adding them to the local hangupcauses viaast_channel_hangupcause_hash_set (Reported by Jaco Kroon)
* ASTERISK-25885 - res_pjsip: Race condition between addingcontact 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 missingmarker bits in RTP streams (Reported by Jacek Konieczny)
* ASTERISK-25881 - pbx: Add support for autohints (Reported byJoshua Colp)
* ASTERISK-25854 - No audio after HOLD/RESUME - incorrecta=recvonly in SDP from Asterisk (Reported by Robert McGilvray)
* ASTERISK-25868 - Sorcery "append to category" should allowfilters (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 whenattempting to operate on a hung up channel (Part 2) (Reported byRichard Mudgett)
* ASTERISK-25867 - [patch] Video delay on app_echo (Reported byJacek Konieczny)
* ASTERISK-24605 - res_parking option parkeddynamic does not workwith the core Features 'parkcall' (DTMF initiated parking)(Reported by Philip Correia)
* ASTERISK-24596 - Unclear how to use Park application withres_parking 'parkeddynamic' enabled. Documentation? (Reported byPhilip Correia)
* ASTERISK-25825 - Crashes during shutdown when running CLIcommands (Reported by Mark Michelson)
* ASTERISK-24543 - Asterisk 13 responds to SIP Invite with allpossible codecs configured for peer as opposed to intersectionof configured codecs and offered codecs (Reported by TaylorHawkes)
* ASTERISK-25407 - Asterisk fails to log to multiple syslogdestinations (Reported by Elazar Broad)
* ASTERISK-25510 - [patch]Log to syslog failing (Reported byMichael Newton)
* ASTERISK-25857 - func_aes: incorrect use of strlen() leads todata corruption (Reported by Gianluca Merlo)
* ASTERISK-25849 - chan_pjsip: transfers with direct mediasometimes 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 inupdate_provisional_keepalive (Reported by Arnd Schmitter)
* ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Localchannel (Reported by Filip Frank)
* ASTERISK-25829 - res_pjsip: PJSIP does not accept spaces whenseparating multiple AORs (Reported by Mateusz Kowalski)
* ASTERISK-25771 - ARI:Crash - Attended transfers of channels intoStasis application. (Reported by Javier Riveros )
* ASTERISK-25830 - Revision 2451d4e breaks NAT (Reported by SeanBright)
* ASTERISK-25582 - Testsuite: Reactor timeout error intests/fax/pjsip/directmedia_reinvite_t38 (Reported by MattJordan)
* ASTERISK-25811 - Unable to delete object from sorcery cache(Reported by Ross Beer)
* ASTERISK-25800 - [patch] Calculate talktime when is first callanswered (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25727 - RPM build requires OPTIONAL_API cflag due toPJSIP requirement (Reported by Gergely Dömsödi)
* ASTERISK-25337 - Crash on PJSIP_HEADER Add P-Asserted-Identitywhen calling from Gosub (Reported by Jacques Peacock)
* ASTERISK-25738 - res_pjsip_pubsub: Crash while executingOutboundSubscriptionDetail ami action (Reported by KevinHarwell)
* ASTERISK-25721 - [patch] res_phoneprov: memory leak andheap-use-after-free (Reported by Badalian Vyacheslav)
* ASTERISK-25272 - [patch]The ICONV dialplan function sometimesreturns garbage (Reported by Etienne Lessard)
* ASTERISK-25751 - res_pjsip: Supportpjsip_dlg_create_uas_and_inc_lock (Reported by Joshua Colp)
* ASTERISK-25606 - Core dump when using transports in sorcery(Reported by Martin Moučka)
* ASTERISK-20987 - non-admin users, who join muted conference arenot being muted (Reported by hristo)
* ASTERISK-25737 - res_pjsip_outbound_registration: line optionnot in Alembic (Reported by Joshua Colp)
* ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEASTVulnerability - Investigate vulnerability of HTTP server(Reported by Alex A. Welzl)
* ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs inudptl_rx_packet cause ast_frdup crash (Reported by WalterDoekes)
* ASTERISK-25742 - Secondary IFP Packets can result in accessinguninitialized pointers and a crash (Reported by Torrey Searle)
* ASTERISK-25397 - [patch]chan_sip: File descriptor leak withnon-default timert1 (Reported by Alexander Traud)
* ASTERISK-25702 - PjSip realtime DB and Cache Errors sinceupgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported byNic Colledge)
* ASTERISK-25730 - build: make uninstall after make distcleantries to remove root (Reported by George Joseph)
* ASTERISK-25725 - core: Incorrect XML documentation may result inweird behavior (Reported by Joshua Colp)
* ASTERISK-25722 - ASAN & testsute: stack-buffer-overflow insip_sipredirect (Reported by Badalian Vyacheslav)
* ASTERISK-25709 - ARI: Crash can occur due to race condition whenattempting to operate on a hung up channel (Reported by MarkMichelson)
* ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reportedby Badalian Vyacheslav)
* ASTERISK-25685 - infrastructure: Run alembic in Jenkins buildscript (Reported by Joshua Colp)
* ASTERISK-25712 - Second call to already-on-call phone andAsterisk 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) reportincorrect values (Reported by Gianluca Merlo)
* ASTERISK-25611 - core: threadpool thread_timeout_thrash unittest sporadically failing (Reported by Joshua Colp)
* ASTERISK-25686 - PJSIP: qualify_timeout is a double, databaseschema 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 duringa transfer (Reported by Kevin Harwell)
* ASTERISK-25697 - bridge_basic: don't play an attended transferfail sound after target hangs up (Reported by Kevin Harwell)
* ASTERISK-25683 - res_ari: Asterisk fails to start if compiledwith MALLOC_DEBUG (Reported by yaron nahum)
* ASTERISK-24097 - Documentation - CHANNEL function help textmissing 'linkedid' argument (Reported by Steven T. Wheeler)
* ASTERISK-25690 - Hanging up when executing connected line subdoes not cause hangup (Reported by Joshua Colp)
* ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'mohreload' cause a crash (Reported by Sean Bright)
* ASTERISK-25632 - res_pjsip_sdp_rtp: RTP is sent from wrong IPaddress when multihomed (Reported by Olivier Krief)
* ASTERISK-25637 - Multi homed server using wrong IP (Reported byDaniel Journo)
* ASTERISK-25394 - pbx: Incorrect device and presence state whenchanging hint details (Reported by Joshua Colp)
* ASTERISK-25640 - pbx: Deadlock on features reload and statechange 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 atshutdown (Reported by Corey Farrell)
* ASTERISK-25679 - res_calendar leaks scheduler. (Reported byCorey Farrell)
* ASTERISK-25675 - Endpoint not listed as Unreachable (Reported byDaniel Journo)
* ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reportedby Corey Farrell)
* ASTERISK-25673 - res_crypto leaks CLI entries (Reported by CoreyFarrell)
* ASTERISK-25668 - res_pjsip: Deadlock in distributor (Reported byMark 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 ofADD_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 aresent for every status change (Reported by George Joseph)
* ASTERISK-25641 - bridge: GOTO_ON_BLINDXFR doesn't work ontransfer initiated channel (Reported by Dmitry Melekhov)
* ASTERISK-25614 - DTLS negotiation delays (Reported by DadeBrandon)
* ASTERISK-25625 - res_sorcery_memory_cache: Add full backendcaching (Reported by Joshua Colp)
* ASTERISK-25601 - json: Audit reference usage and thread safety(Reported by Joshua Colp)
* ASTERISK-25624 - AMI Event OriginateResponse bug (Reported bysungtae 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 neverupdated in wait_our_turn function (app_queue.c) (Reported byCarlos Oliva)
* ASTERISK-25364 - [patch]Issue a TCP connection(kernel) andthread of asterisk is not released (Reported by Hiroaki Komatsu)
* ASTERISK-25569 - app_meetme: Audio quality issues (Reported byCorey Farrell)
* ASTERISK-25619 - res_chan_stats not sending the correctinformation to StatsD (Reported by Tyler Cambron)
* ASTERISK-24146 - [patch]No audio on WebRtc caller side whenanswer waiting time is more than ~7sec (Reported by AlekseiKulakov)
* ASTERISK-25609 - [patch]Asterisk may crash when callingast_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 PLCwith FEC (Reported by Alexander Traud)
* ASTERISK-25610 - Asterisk crash during "sip reload" (Reported byDudás József)
* ASTERISK-25608 - res_pjsip/contacts/statsd: Lifecycle eventsaren'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 (OpusCodec) (Reported by Alexander Traud)
* ASTERISK-25498 - Asterisk crashes when negotiating g729 withoutthat module installed (Reported by Ben Langfeld)
* ASTERISK-25595 - Unescaped : in messge sent to statsd (Reportedby Niklas Larsson)
* ASTERISK-25598 - res_pjsip: Contact status messages areprinting a hash instead of the uri (Reported by George Joseph)
* ASTERISK-25600 - bridging: Inconsistency in BRIDGEPEER (Reportedby Jonathan Rose)
* ASTERISK-25476 - chan_sip loses registrations after a while(Reported by Michael Keuter)
* ASTERISK-25593 - fastagi: record file closed after sendingresult (Reported by Kevin Harwell)
* ASTERISK-25585 - [patch]rasterisk never hits most of main(), butit'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 registrationscreated via ARI are not loaded into memory on Asteriskstart/restart (Reported by Matt Jordan)
* ASTERISK-25545 - [patch] translation module gets cached notjoint format (Reported by Alexander Traud)
* ASTERISK-25573 - [patch] H.264 format attribute module: resetswhole SDP (Reported by Alexander Traud)
* ASTERISK-24958 - Forwarding loop detection inhibits certaindesirable 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 AlecDavis)
* ASTERISK-25565 - DNS: System resolver only returns 1 record perresult (Reported by George Joseph)
* ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported byJoshua Colp)
* ASTERISK-25160 - [patch] Opus Codec: SIP/SDP line fmtp missingwhen called internally (Reported by Alexander Traud)
* ASTERISK-25535 - [patch] format creation on module load insteadof cache (Reported by Alexander Traud)
* ASTERISK-25449 - main/sched: Regression introduced by5c713fdf18f causes erroneous duplicate RTCP messages; otherpotential scheduling issues in chan_sip/chan_skinny (Reported byMatt Jordan)
* ASTERISK-25546 - threadpool: Race condition between idle timeoutand activation (Reported by Joshua Colp)
* ASTERISK-25537 - [patch] format-attribute module: RFC orinternal defaults? (Reported by Alexander Traud)
* ASTERISK-25533 - [patch] buffer for ast_format_cap_get_namesonly 64 bytes (Reported by Alexander Traud)
* ASTERISK-25373 - add documentation for CALLERID(pres) and alsothe CONNECTEDLINE and REDIRECTING variants (Reported by WalterDoekes)
* ASTERISK-25528 - DNS: System resolver issues with TTL parse(Reported by dtryba)
* ASTERISK-25527 - Quirky xmldoc description wrapping (Reported byWalter Doekes)
* ASTERISK-24779 - Passthrough OPUS codec not working withchan_pjsip (Reported by PowerPBX)
* ASTERISK-25522 - ARI: Crash when creating channel via ARIoriginate with requesting channel (Reported by Matt Jordan)
* ASTERISK-25434 - Compiler flags not reported in 'core showsettings' despite usage during compilation (Reported by RustyNewton)
* ASTERISK-24106 - WebSockets Automatically decides what driver itwill use (Reported by Andrew Nagy)
* ASTERISK-25513 - Crash: malloc failed with high load ofsubscriptions. (Reported by John Bigelow)
* ASTERISK-25505 - res_pjsip_pubsub: Crash on off-nominal when UASdialog can't be created (Reported by Joshua Colp)
* ASTERISK-25494 - build: GCC 5.1.x catches some new const, arraybounds and missing paren issues (Reported by George Joseph)
* ASTERISK-25485 - res_pjsip_outbound_registration: registrationstops 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 valuesin frame.c (Reported by dea)
* ASTERISK-25484 - [patch] autoframing=yes has no effect (Reportedby Alexander Traud)
* ASTERISK-25308 - ari: Websocket leak (Reported by Joshua Colp)
* ASTERISK-25461 - Nested dialplan #includes don't work asexpected. (Reported by Richard Mudgett)
* ASTERISK-25455 - Deadlock of PJSIP realtime overres_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 greatlyexceeds zero. (Reported by Dmitriy Serov)
* ASTERISK-25451 - Broken video - erased rtp marker bit (Reportedby Stefan Engström)
* ASTERISK-25400 - Hints broken when "CustomPresence" doesn'texist in AstDB (Reported by Andrew Nagy)
* ASTERISK-25443 - [patch]IPv6 - Potential issue in via headerparsing (Reported by ffs)
* ASTERISK-25404 - segfault/crash in chan_pjsip_hangup ... atchan_pjsip.c (Reported by Chet Stevens)
* ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON(Reported by Bojan Nemčić)
* ASTERISK-25441 - Deadlock in res_sorcery_memory_cache. (Reportedby Richard Mudgett)
* ASTERISK-25438 - res_rtp_asterisk: ICE role message even whenICE is not enabled (Reported by Joshua Colp)
* ASTERISK-25383 - Core dumps on startup and shutdown withMALLOC_DEBUG enabled (Reported by yaron nahum)
* ASTERISK-25423 - Caller gets no Connected line update duringcall pickup. (Reported by Richard Mudgett)
* ASTERISK-25305 - Dynamic logger channels can be added multipletimes (Reported by Mark Michelson)
* ASTERISK-25418 - On-hold channels redirected out of a bridgeappear to still be on hold (Reported by Mark Michelson)
* ASTERISK-25384 - Regular Asterisk crashes when using Pageapplication. "user_data is NULL" (Reported by Chet Stevens)
* ASTERISK-25410 - app_record: RECORDED_FILE variable not beingpopulated (Reported by Kevin Harwell)
* ASTERISK-25396 - chan_sip: Extremely long callerid name causesinvalid 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 Framesize = 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 certainconfiguration backends (Reported by Mark Michelson)
* ASTERISK-25387 - res_pjsip_nat: Malformed REGISTER requestcauses NAT'd Contact header to not be rewritten (Reported byMatt Jordan)
* ASTERISK-25227 - No audio at in-band announcements in ooh323channel (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 othermechanism) do not destroy their related contacts (Reported byMatt Jordan)
* ASTERISK-25369 - res_parking: ParkAndAnnounce - Inheritablevariables aren't applied to the announcer channel (Reported byJonathan Rose)
* ASTERISK-25367 - pbx: Long pattern match hints may cause "coreshow hints" to crash (Reported by Joshua Colp)
* ASTERISK-25365 - Persistent subscriptions have extraContent-Length/corrupted messages (Reported by Mark Michelson)
* ASTERISK-25356 - res_pjsip_sdp_rtp: Multiple keepalive scheduleditems may exist (Reported by Joshua Colp)
* ASTERISK-25355 - sched: ast_sched_del may return prematurely dueto 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 hangupcause on call pickup (Reported by Joshua Colp)
* ASTERISK-25342 - res_pjsip: Repeated usage of pj_gethostip mayblock (Reported by Joshua Colp)
* ASTERISK-25341 - bridge: Hangups may get lost when executingactions (Reported by Joshua Colp)
* ASTERISK-25339 - res_pjsip: Empty "auth" sections fromnon-config backgrounds are interpreted as valid (Reported byMatt Jordan)
* ASTERISK-25215 - Differences in queue.log between SetQUEUE_MEMBER and using PauseQueueMember (Reported by LorneGaetz)
* ASTERISK-25322 - Crash occurs when using MixMonitor with t() orr() options. (Reported by Richard Mudgett)
* ASTERISK-25320 - chan_sip.c: sip_report_security_event searchesfor wrong or non existent peer on invite (Reported by KevinHarwell)
* ASTERISK-25312 - res_http_websocket: Terminate connection onfatal cases (Reported by Joshua Colp)
* ASTERISK-25315 - DAHDI channels send shortened duration DTMFtones. (Reported by Richard Mudgett)
* ASTERISK-25306 - Persistent subscriptions can save multiple SIPmessages at once, leading to potential crashes. (Reported byMark Michelson)
* ASTERISK-25309 - [patch] iLBC 20 advertised (Reported byAlexander Traud)
* ASTERISK-25304 - res_pjsip: XML sanitization may write pastbuffer (Reported by Joshua Colp)
* ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer onFirefox 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 applicationduring WebSocket creation don't receive a StasisStart event(Reported by Matt Jordan)
* ASTERISK-25296 - RTP performance issue with several channeldrivers. (Reported by Richard Mudgett)
* ASTERISK-25297 - Crashes runningchannels/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 channelnot 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 hearringback (Reported by Etienne Lessard)
* ASTERISK-25253 - confbridge volume options and other volumecontrols such as func_volume don't work (Reported by DmitriySerov)
* 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 failuredue to reversed if logic (Reported by Elazar Broad)
* ASTERISK-24867 - Docs for 'e' option in ResetCDR say to useCDR_PROP instead, CDR_PROP docs are unclear (Reported by RustyNewton)
* ASTERISK-24853 - Documentation claims chan_sip outboundregistrations support WS or WSS as valid transports (not true)(Reported by PSDK)
* ASTERISK-25242 - PJSIP: No audio when Asterisk inside NAT andendpoints outside NAT - implement functionality similar tochan_sip 'rtpkeepalive'? (Reported by Mark Michelson)
* ASTERISK-25258 - chan_pjsip: Incorrect format switch on receivedRTP packet (Reported by Joshua Colp)
* ASTERISK-25257 - [patch]channels/sig_pri.h -> sig_pri_span ->force_restart_unavailable_chans in wrong scope (Reported byPatric Marschall)
* ASTERISK-24934 - [patch]Asterisk manager output does not escapecontrol characters (Reported by warren smith)
* ASTERISK-25255 - Missing AMI VarSet events when setting to anempty string. (Reported by Richard Mudgett)
* ASTERISK-25254 - Crash if dialplan sets ATTENDEDTRANSFER to anempty string before Park. (Reported by Richard Mudgett)
* ASTERISK-25183 - PJSIP: Crash on NULL channel inchan_pjsip_incoming_response despite previous checks for NULLchannel (Reported by Matt Jordan)
* ASTERISK-25201 - Crash in PJSIP distributor on already free'dthreadpool (Reported by Matt Jordan)
* ASTERISK-25240 - bridge_native_rtp: Direct media wrongfullystarted when completing attended transfer (Reported by JoshuaColp)
* ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes(Reported by Rusty Newton)
* ASTERISK-25146 - DNS: Create system level resolver (Reported byJoshua Colp)
* ASTERISK-22805 - res_rtp_asterisk: Crash when callingBIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP(Reported by Dmitry Burilov)
* ASTERISK-24550 - res_rtp_asterisk: Crash inast_rtp_on_ice_complete during DTLS handshake (Reported byOsaulenko Alexander)
* ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported byBadalian Vyacheslav)
* ASTERISK-24832 - [patch]DTLS-crashes within openssl (Reportedby Stefan Engström)
* ASTERISK-25127 - DTLS crashes following "Unable to cancelschedule ID" in dtls_srtp_check_pending (Reported by DadeBrandon)
* ASTERISK-25168 - Random Core Dumps on Asterisk 13.4 PJSIP, inast_channel_name at channel_internal_api.c (Reported by CarlFortin)
* ASTERISK-25076 - res_pjsip: Failover does not occur onconnection-less transport or 503 response (Reported by JoshuaColp)
* ASTERISK-25226 - chan_sip: Channel leak in branch 13 on earlyreplaces call pickup (Reported by Walter Doekes)
* ASTERISK-25222 - Crash in recurring cancel callback called fromast_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 memcpyin 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 and13.4 (Reported by cervajs)
* ASTERISK-25196 - res_pjsip_nat: rewrite_contact should not beapplied to Contact header when Record-Route headers are present(Reported by Mark Michelson)
* ASTERISK-24907 - res_pjsip_outbound_registration: crash duringunload if registration attempts are still occuring (Reported byKevin Harwell)
* ASTERISK-25204 - res_pjsip_refer: Duplicated Referred-By orReplaces headers on outbound INVITEs. (Reported by MarkMichelson)
* ASTERISK-25189 - AMI: Add Linkedid header to standard channelsnapshot information. (Reported by Richard Mudgett)
* ASTERISK-25171 - Early completion of feature code attendedtransfer results in intermittent one-way audio, "ghost ringing"and robotic sound. (Reported by Rusty Newton)
* ASTERISK-25172 - Crash in channels/sip/sip blindtransfer/caller_refer_only test inast_format_cap_append_from_cap during ast_request (Reported byMatt Jordan)
* ASTERISK-25180 - res_pjsip_mwi: Unsolicited MWI requires reload(Reported by Joshua Colp)
* ASTERISK-25182 - [patch] on CLI sip reload, new codecs getappended only (Reported by Alexander Traud)
* ASTERISK-25163 - Deadlock in chan_sip between reload of sip peercontainer and MWI Stasis callback (Reported by Dmitriy Serov)
* ASTERISK-25091 - Asterisk REST API - bridge.addChannel crashasterisk 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 whennegotiating g.726 (Reported by Kevin Harwell)
* ASTERISK-24344 - CDR_PROP(disable) disables CDR only for firstdialed party (Reported by Janusz Karolak)
* ASTERISK-24443 - CDR fields (dst, dcontext) empty in transfercall started from Macro (Reported by Arveno Santoro)
* ASTERISK-25154 - [patch]fromtag may need to be updated aftersuccessful call dialog match (Reported by Damian Ivereigh)
* ASTERISK-25156 - chan_pjsip’s CHAN_START cel event lacks thecorrect context and exten (Reported by cloos)
* ASTERISK-25157 - bridging: Performing a blonde transfer does notresult in connected line updates (Reported by Joshua Colp)
* ASTERISK-25087 - Asterisk segfault when using Directoryapplication with alias option and specific mailbox configuration(Reported by Chet Stevens)
* ASTERISK-25115 - Crash related to funcsip_resolve_invoke_user_callback of res_pjsip/pjsip_resolver.c(Reported by John Bigelow)
* ASTERISK-25096 - [patch]Segfault when registering overwebsockets 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 attributebut 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 "standardsystem utilities" (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25148 - res_pjsip NULL channel audit (Reported by MarkMichelson)
* ASTERISK-25131 - chan_pjsip: In-dialog authentication nothandled. (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 addressthat end with ::80 (Reported by Mark Petersen)
* ASTERISK-25122 - Large SIP packet received via pjsip overwebsocket crashes Asterisk (Reported by Ivan Poddubny)
* ASTERISK-25121 - Stasis: Fix unsafe use of stasis_unsubscribe inmodules. (Reported by Corey Farrell)
* ASTERISK-25120 - Astobj2: Weakproxy subscriptions should be runin reverse order. (Reported by Corey Farrell)
* ASTERISK-25105 - res_pjsip: Possible incompatibility betweenqualify_timeout and pjproject-2.4 (Reported by George Joseph)
* ASTERISK-25117 - res_mwi_external_ami: Fix manager actionregistrations. (Reported by Corey Farrell)
* ASTERISK-25112 - Logger: Configuration settings are not reset todefault during reload. (Reported by Corey Farrell)
* ASTERISK-24983 - IAX deadlock between hangup and scheduledactions (ex. largrq) (Reported by Y Ateya)
* ASTERISK-24944 - main/audiohook.c change prevents G722 callrecording (Reported by Ronald Raikes)
* ASTERISK-25110 - res_resolver_unbound.c compilation failure:SIGURG is undeclared in func unbound_resolver_stop (Reported byJohn Bigelow)
* ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2or more digits (Reported by Makoto Dei)
* ASTERISK-25086 - [patch]PJSIP crashes if endpoint missing inDial() (Reported by snuffy)
* ASTERISK-25089 - res_pjsip_config_wizard: Variable specified intemplates aren't being processed correctly (Reported by GeorgeJoseph)
* ASTERISK-25090 - CLI core show channel truncates cdr variables(Reported by snuffy)
* ASTERISK-25083 - Message.c: Message channel becomes saturatedwith frames leading to spammy log messages (Reported by JonathanRose)
* ASTERISK-25085 - [patch]Potential crash after unload offunc_periodic_hook or test_message (Reported by Corey Farrell)
* ASTERISK-25082 - Asterisk deletes message after doing a playbackof an INBOX message using ast_vm_play when the Old folder isfull for that mailbox. (Reported by Jonathan Rose)
* ASTERISK-21893 - Segfault after call hangup, inast_channel_hangupcause_set, at channel_internal_api.c (Reportedby Aleksandr Gordeev)
* ASTERISK-25042 - asterisk.conf options override command-lineoptions. (Reported by Corey Farrell)
* ASTERISK-25074 - Regression: Recent clang-related change brokecross compiling of Asterisk (Reported by Sebastian Kemper)
* ASTERISK-24442 - Outgoing call files don't work properly whenset 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 inres_config_mysql addon (Reported by Alexandre Fournier)
* ASTERISK-25057 - res_pjsip_pubsub: Crash in send_notify due toinvalid root pointer in sub_tree (Reported by Matt Jordan)
* ASTERISK-24938 - ARI Snoop Channel results in excessiveescalating CPU usage (Reported by George Ladoff)
* ASTERISK-25034 - chan_dahdi: Some telco switches occasionallyignore ISDN RESTART requests. (Reported by Richard Mudgett)
* ASTERISK-25003 - Asterisk crashes on attended transfer (usingfeature) (Reported by Artem Volodin)
* ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not alwayscontain waiting time (Reported by Etienne Lessard)
* ASTERISK-25027 - Build System: Many ARI modules are missingdependencies. (Reported by Corey Farrell)
* ASTERISK-25061 - pbx_config: Register manager actions withmodule 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 (inast_channel_snapshot_create at stasis_channels.c) with CertifiedAsterisk 13. (Reported by Chet Stevens)
* ASTERISK-25053 - Unit test category /main/presence missingtrailing slash. (Reported by Corey Farrell)
* ASTERISK-22708 - res_odbc.conf negative_connection_cache optionnot respected, failover between DSNs doesn't work (Reported byJoshE)
* ASTERISK-25054 - Formats interface's cannot be unregistered,needs to hold modules until shutdown. (Reported by CoreyFarrell)
* 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 withoutPJSip (Reported by Peter Whisker)
* ASTERISK-24896 - [patch] Using force black background leads tocolours not being reset (Reported by dant)
* ASTERISK-25048 - Astobj2: Initialization order wrong when bothrefdebug and AO2_DEBUG are both enabled. (Reported by CoreyFarrell)
* ASTERISK-19608 - Asterisk-1.8.x starts rejecting calls withcause code 44 after some time. (Reported by Denis AlbertoMartinez)
* ASTERISK-25037 - res_pjsip_outbound_registration: Potentialcrash in off-nominal failure case when sending message (Reportedby Joshua Colp)
* ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls(Reported by Steve Davies)
* ASTERISK-22790 - check_modem_rate() may return incorrect ratefor V.27 (Reported by not here)
* ASTERISK-23231 - Since 405693 If we have res_fax.conf file setto minrate=2400, then res_fax refuse to load (Reported by DavidBrillert)
* ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,which is disallowed in res_fax's check_modem_rate (Reported byMatt Jordan)
* ASTERISK-25020 - Mismatched response to outgoing REGISTERrequest (Reported by Mark Michelson)
* ASTERISK-25028 - Build System: Unneeded defines inasterisk/buildopts.h (Reported by Corey Farrell)
* ASTERISK-25026 - Git conversion: Non-C files not switched toASTERISK_REGISTER_FILE (Reported by Corey Farrell)
* ASTERISK-24996 - chan_pjsip: Creating Channel Causes Asterisk toCrash When Duplicate AOR Sections Exist in pjsip.conf (Reportedby Ashley Sanders)
* ASTERISK-25018 - pjsip show endpoints crashes asterisk whenqualified aors present (Reported by Ivan Poddubny)
* ASTERISK-24749 - ConfBridge: Wrong language on playingconf-hasjoin and conf-hasleft when played to bridge (Reported byPhilippe Bolduc)
* ASTERISK-24845 - pjsip send notify not working with Cisco phone(Reported by Carl Fortin)
* ASTERISK-25004 - Crash in authenticated reinvite afteroriginated 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 evenif res_corosync.conf is missing (Reported by George Joseph)
* ASTERISK-24997 - Astobj2: Some callers of __adjust_lock do notpre-check the object (Reported by Corey Farrell)
* ASTERISK-24994 - dns: Query set unit tests are failing due torace condition (Reported by Joshua Colp)
* ASTERISK-24982 - res_pjsip_mwi: Unsolicited MWI NOTIFY only senton 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 noHangupRequest event comes for the dahdi channel. (Reported byAndrew Zherdin)
* ASTERISK-24977 - Contacts that don't use qualify are beingmarked as unavailable (Reported by George Joseph)
* ASTERISK-24774 - Segfault in ast_context_destroy withextensions.ael and extensions.conf (Reported by Corey Farrell)
* ASTERISK-24841 - ConfBridge: Strange sampling rates chosen whenchannels have multiple native formats (Reported by Matt Jordan)
* ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Buildto Fail (Reported by Ashley Sanders)
* ASTERISK-24863 - res_pjsip: No endpoint events raised via AMIwhen contacts cannot be reached/qualified (Reported by DmitriySerov)
* ASTERISK-24869 - Asterisk segfaults on DAHDI attended transferdue to application (appl) being NULL on unbridged channel(Reported by viniciusfontes)
* ASTERISK-24970 - Crash in res_pjsip_pubsub handling of failednotify (Reported by Scott Griepentrog)
* ASTERISK-13271 - menuselect sets defaults too late (Reported byJohn Nemeth)
* ASTERISK-24959 - [patch]CLI command cdr show pgsql status(Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-20524 - AMI improperly handles lines of exactly 1025characters (Reported by David M. Lee)
* ASTERISK-24936 - New Feature: AO2 weakproxy objects (Reported byCorey Farrell)
* ASTERISK-24954 - Git migration: Asterisk version numbers areincompatible 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 nothonored (Reported by Juergen Spies)
* ASTERISK-24835 - Early Media Not working with Chan SIP andAsterisk 13 (Reported by Andrew Nagy)
* ASTERISK-21777 - Asterisk tries to transcode video instead ofaudio (Reported by Nick Ruggles)
* ASTERISK-24380 - core: Native formats are set to h264 withcertain audio/video codec configuration, resulting in pathtranslation WARNINGs (Reported by Matt Jordan)
* ASTERISK-22352 - [patch] IAX2 custom qualify timer is not takeninto account (Reported by Frederic Van Espen)
* ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer tooshort (Reported by Y Ateya)
* ASTERISK-24935 - res_pjsip_phoneprov_provider: Fix leakedOBJ_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 JonathanRose)
* ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULLbyte prefix bug (Reported by Matt Jordan)
* ASTERISK-21211 - chan_iax2 - unprotected access ofiaxs[peer->callno] potentially results in segfault (Reported byJaco Kroon)
* ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working(Reported by Christoph Timm)
* ASTERISK-24910 - "timer=no" and "timer=required" settings inpjsip.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 ofvoicemail 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 besent out of order (Reported by Mark Michelson)
* ASTERISK-24920 - Asterisk handles duplicate SIP requests as ifthey were each a new request (Reported by Mark Michelson)
* ASTERISK-24781 - PJSIP: Unnecessary 180 Ringing messages sentwith undesireabe consequences. (Reported by Richard Mudgett)
* ASTERISK-24857 - [patch] "timing test", pjsip incoming/outgoingcalls, voicemail prompts and recordings all fail when using thekqueue timer source on FreeBSD 10.x (Reported by Justin T.Gibbs)
* ASTERISK-24155 - [patch]Non-portable and non-reliable recursiondetection in ast_malloc (Reported by Timo Teräs)
* ASTERISK-24142 - CCSS: crash during shutdown due to devicelookup in destroyed container (Reported by David Brillert)
* ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal duringcore restart now (Reported by Peter Katzmann)
* ASTERISK-24805 - [patch] - ASAN: Race condition(heap-use-after-free) on asterisk closing (Reported by BadalianVyacheslav)
* ASTERISK-24881 - ast_register_atexit should only be used whenabsolutely needed (Reported by Corey Farrell)
* ASTERISK-24731 - res_pjsip_session cannot be unloaded (Reportedby Corey Farrell)
* ASTERISK-24864 - app_confbridge: file playback blocks dtmf(Reported by Kevin Harwell)
* ASTERISK-14233 - [patch] Buddies are always auto-registered whenprocessing the roster (Reported by Simon Arlott)
* ASTERISK-24780 - [patch] - Buddies are always auto-registeredwhen processing the roster (Reported by Simon Arlott)
* ASTERISK-24879 - [patch]Compilation fails due to 64bit timeunder OpenBSD (Reported by snuffy)
* ASTERISK-24880 - [patch]Compilation under OpenBSD (Reported bysnuffy)
* ASTERISK-21765 - [patch] - FILE function's length argumentcounts from beginning of file rather than the offset (Reportedby John Zhong)
* ASTERISK-24817 - init_logger_chain: unreachable code block(Reported by Corey Farrell)
* ASTERISK-24882 - chan_sip: Improve usage of REF_DEBUG (Reportedby Corey Farrell)
* ASTERISK-24876 - Investigate reference leaks fromtests/channels/local/local_optimize_away (Reported by CoreyFarrell)
* ASTERISK-24840 - res_pjsip: conflicting endpoint identifiers(Reported by Kevin Harwell)
* ASTERISK-16779 - Cannot disallow unknown format '' (Reported byAtis Lezdins)
* ASTERISK-18708 - func_curl hangs channel under load (Reported byDave Cabot)
* ASTERISK-21038 - Bad command completion of "core set debugchannel" (Reported by Richard Kenner)
* ASTERISK-19470 - Documentation on app_amd is incorrect (Reportedby Frank DiGennaro)
* ASTERISK-24872 - [patch] AMI PJSIPShowEndpoint closes AMIconnection 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 thesame/similar functionality. (Reported by Diederik de Groot)
* ASTERISK-24807 - Missing mandatory field Max-Forwards (Reportedby Anatoli)
* ASTERISK-24808 - res_config_odbc: Improper escaping ofbackslashes occurs with MySQL (Reported by Javier Acosta)
* ASTERISK-23390 - NewExten Event with application AGI shows upbefore and after AGI runs (Reported by Benjamin Keith Ford)
* ASTERISK-24786 - [patch] - Asterisk terminates when playing avoicemail 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 totransferrer during attended transfer when using a Stasis bridge(Reported by John Bigelow)
* ASTERISK-24830 - res_rtp_asterisk.c checks USE_PJPROJECT notHAVE_PJPROJECT (Reported by Stefan Engström)
* ASTERISK-24825 - Caller ID not recognized usingCentrex/Distinctive dialing (Reported by Richard Mudgett)
* ASTERISK-17588 - Caller ID on TDM410P *UK* PSTN (Reported byDaniel Flounders)
* ASTERISK-24838 - chan_sip: Locking inversion occurs whenbuilding a peer causes a peer poke during request handling(Reported by Richard Mudgett)
* ASTERISK-24751 - Integer values in json payload to ARI causeasterisk to crash (Reported by jeffrey putnam)
* ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
* ASTERISK-18105 - most of asterisk modules are unbuildable incygwin environment (Reported by feyfre)
* ASTERISK-21845 - maxcalls exceeded, Asterisk sends out 480 andalso BYE (Reported by Tony Ching)
* ASTERISK-15434 - [patch] When ast_pbx_start failed, both anerror response and BYE are sent to the caller (Reported byMakoto Dei)
* ASTERISK-23214 - chan_sip WARNING message 'We are requestingSRTP for audio, but they responded without it' is ambiguous andwrong in some cases (Reported by Rusty Newton)
* ASTERISK-17721 - Incoming SRTP calls that specify a key lifetimefail (Reported by Terry Wilson)
* ASTERISK-20233 - SRTP not working with some devices (EgGrandstream gxv3175) - Message "Can't provide secure audiorequested 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 invalidthread ID being passed to pthread_kill (Reported by JoshE)
* ASTERISK-24812 - ARI: Creating channels through /channelsresource 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 unhelpfulresponse on non-existent variable (Reported by Joshua Colp)
* ASTERISK-24785 - 'Expires' header missing from 200 OK onREGISTER (Reported by Ross Beer)
* ASTERISK-24499 - Need more explicit debug when PJSIP dialstringis 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 moduleunload (Reported by Corey Farrell)
* ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gccOSX with 64 bit integers (Reported by Corey Farrell)
* ASTERISK-24787 - [patch] - Microsoft exchange incompatibilityfor playing back messages stored in IMAP - play_message: Noorigtime (Reported by Graham Barnett)
* ASTERISK-22670 - Asterisk crashes when processing ISDN AoCEvents (Reported by klaus3000)
* ASTERISK-24689 - Segfault on hangup after outgoing PRI-Euroisdncall (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 toSSLv3_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 toast_bridge_transfer_attended() (Reported by Zane Conkle)
* ASTERISK-24791 - Crash in ast_rtcp_write_report (Reported byJoshE)
* ASTERISK-24085 - Documentation - We should remove or furtherdocument the 'contact' section in pjsip.conf (Reported by RustyNewton)
* ASTERISK-24632 - install_prereq script installs pjprojectwithout IPv6 support (Reported by Rusty Newton)
* ASTERISK-24685 - "pjsip show version" CLI command (Reported byJoshua Colp)
* ASTERISK-24768 - res_timing_pthread: file descriptor leak(Reported by Matthias Urlichs)
* ASTERISK-24612 - res_pjsip: No information if a required sorcerywizard is not loaded (Reported by Joshua Colp)
* ASTERISK-24716 - Improve pjsip log messages for presencesubscription failure (Reported by Rusty Newton)
* ASTERISK-24771 - ${CHANNEL(pjsip)} - segfault (Reported byNiklas Larsson)
* ASTERISK-24727 - PJSIP: Crash experienced during multi-Asterisktransfer 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 (Reportedby Zane Conkle)
* ASTERISK-24701 - Stasis: Write timeout on WebSocket fails tofully disconnect underlying socket, leading to events beingdropped with no additional information (Reported by Matt Jordan)
* ASTERISK-24752 - Crash in bridge_manager_service_req when bridgeis destroyed by ARI during shutdown (Reported by RichardMudgett)
* ASTERISK-24772 - ODBC error in realtime sippeers when deviceunregisters 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 inres_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 insorcery.conf false ERROR messages may occur (Reported by JoshuaColp)
* ASTERISK-24616 - Crash in res_format_attr_h264 due to invalidstring copy (Reported by Yura Kocyuba)
* ASTERISK-24737 - When agent not logged in, agent status showsunavailable, queue status shows agent invalid (Reported byRichard Mudgett)
* ASTERISK-24635 - PJSIP outbound PUBLISH crashes when no responseis ever received (Reported by Marco Paland)
* ASTERISK-24736 - Memory Leak Fixes (Reported by Mark Michelson)
* ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported byStephan Eisvogel)
* ASTERISK-24711 - DTLS handshake broken with latest OpenSSLversions (Reported by Jared Biel)
* ASTERISK-24666 - Security Vulnerability: RTP not closed aftersip call using unsupported codec (Reported by Y Ateya)
* ASTERISK-24676 - Security Vulnerability: URL request injectionin libCURL (CVE-2014-8150) (Reported by Matt Jordan)
* ASTERISK-24729 - Outbound registration not occuring on newregistrations after reload. (Reported by Richard Mudgett)
* ASTERISK-24728 - tcptls: Bad file descriptor error whenreloading chan_sip (Reported by Kevin Harwell)
* ASTERISK-24721 - manager: ModuleLoad action incorrectly reports'module not found' during a Reload operation (Reported by MattJordan)
* ASTERISK-24715 - chan_sip: stale nonce causes failure (Reportedby Kevin Harwell)
* ASTERISK-24485 - res_pjsip cannot be unloaded or shutdown(Reported by Corey Farrell)
* ASTERISK-24719 - ConfBridge recording channels get stuck whenrecording started/stopped more than once (Reported by RichardMudgett)
* 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_timedwaitin bridge_channel.c (Reported by George Joseph)
* ASTERISK-24544 - Compile fails on OSX Yosemite because ofincorrect detection of htonll and ntohll (Reported by GeorgeJoseph)
* ASTERISK-24231 - crash: CLI execution of realtime destroysippeers id 1 causes crash due to NULL name provided toast_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 sensitivecolumn comparison for 'defaultuser' (Reported byHZMI8gkCvPpom0tM)
* ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitorm() option does not queue an MWI event (Reported by GarethPalmer)
* ASTERISK-24673 - outgoing sip registers cannot be removed ormodified without doing restart (or doing module unloadchan_sip.so) (Reported by Stefan Engström)
* ASTERISK-24640 - Registration pending stays forever after sipreload (Reported by Max Man)
* ASTERISK-24682 - app_dial: Multiple DialEnd events emitted whenMACRO_RESULT or GOSUB_RESULT are an unexpected value (Reportedby Matt Jordan)
* ASTERISK-24560 - Creating a named ARI bridge twice causes acrash (Reported by Kinsey Moore)
* ASTERISK-24600 - Stuck IAX channels, Asterisk stops respondingto most traffic, potential deadlock (Reported by Jeff Collell)
* ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects32-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 LEIFU)
* ASTERISK-24615 - When Multiple Transports Exist in pjsip.conf,Incorrect External Addresses is Used in SIP Packets WhenResponding to INVITE (Reported by David Justl)
* ASTERISK-24624 - Transfer to invalid extension results in hungchannel. (Reported by Zane Conkle)
* ASTERISK-24663 - [patch] Unnamed semaphore autoconf check failson cross compilation (Reported by abelbeck)
* ASTERISK-24655 - res_pjsip_outbound_publish: Hang on shutdownwhile attempting to publish (Reported by Kevin Harwell)
* ASTERISK-23991 - [patch]asterisk.pc file contains a small errorin the CFlags returned (Reported by Diederik de Groot)
* ASTERISK-23850 - Park Application does not respect ReturnContext Priority (Reported by Andrew Nagy)
* ASTERISK-24665 - Configure check required forpjsip_get_dest_info() (Reported by Mark Michelson)
* ASTERISK-24049 - Asterisk Manager Interface: A number of listtype responses aren't using astman_send_listack (Reported byJonathan Rose)
* ASTERISK-20744 - [patch] Security event logging does not workover 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 doesnot 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 channelcannot be Redirected (Reported by Kinsey Moore)
* ASTERISK-24376 - res_pjsip_refer: REFER request for remotesession attempts to direct channel to external_replacesextension instead of context, without providing for theReferred-To SIP URI (Reported by Matt Jordan)
* ASTERISK-24513 - Local channel apparently leaked in off-nominalDTMF attended transfer (Reported by Mark Michelson)
* ASTERISK-24367 - PJSIP: allow all results in failure to sendINVITE (Reported by Scott Griepentrog)
* ASTERISK-24267 - Queue variables associated withsetinterfacevar, setqueueentryvar, setqueuevar are not passed tolocal channel (Reported by Mitch Claborn)
* ASTERISK-24641 - Deadlock in Trunk (Reported by MalcolmDavenport)
* ASTERISK-23841 - DTMF atxfer doesn't set CallerID for the recallcalls to the transferrer. (Reported by Richard Mudgett)
* ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrongdestination when 'sendrpid=yes' (in proxy environment) (Reportedby Karsten Wemheuer)
* ASTERISK-23733 - 'reload acl' fails if acl.conf is not presenton startup (Reported by Richard Kenner)
* ASTERISK-24566 - Uninit buf in WS write (Reported by BadalianVyacheslav)
* ASTERISK-24337 - Spammy DEBUG message needs to be at a higherlevel - 'Remote address is null, most likely RTP has beenstopped' (Reported by Rusty Newton)
* ASTERISK-24459 - bridge_native_rtp: Native RTP bridging ischosen for RTP compatible channels when the DTMF mode is notcompatible (Reported by Yaniv Simhi)
* ASTERISK-24536 - AMI redirect with PJSIP fails to move extrachannel (Reported by Niklas Larsson)
* ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wronglycasts char to unsigned int (Reported by Walter Doekes)
* ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP isenabled (Reported by Andreas Steinmetz)
* ASTERISK-22455 - Asterisk 12 on Ubuntu Lucid deadlocks withDEBUG_THREADS+OPTIONAL_API enabled (Reported by David M. Lee)
* ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flagenabled (Reported by Richard Mudgett)
* ASTERISK-24604 - res_rtp_asterisk: Crash during restart due torace condition in accessing codec in stored ast_frame and codeccore (Reported by Matt Jordan)
* ASTERISK-24563 - Direct Media calls within private networksometimes get one way audio (Reported by Kevin Harwell)
* ASTERISK-24607 - res_pjsip_session: re-INVITE with declinedmedia streams results in 488 (Reported by Matt Jordan)
* ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSSfrom JSSIP (Reported by Badalian Vyacheslav)
* ASTERISK-24514 - res_pjsip_outbound_registration: stack overflowwhen using non-default sorcery wizard (Reported by KevinHarwell)
* ASTERISK-24342 - PJSIP: Qualifying endpoints attempts to do themall at the same time. (Reported by Richard Mudgett)
* ASTERISK-24556 - Asterisk 13 core dumps when calling from pjsipextension to another pjsip extension (Reported by Abhay Gupta)
* ASTERISK-24537 - Stasis: StasisStart/StasisEnd events are notreliably transmitted during transfers (Reported by Matt Jordan)
* ASTERISK-24573 - [patch]Out of sync conversation recording whendivided in multiple recordings (Reported by Nuno Borges)
* ASTERISK-24572 - [patch]App_meetme is loaded without itsdefaults when the configuration file is missing (Reported byNuno Borges)
* ASTERISK-24516 - [patch]Asterisk segfaults when playing backvoicemail under high concurrency with an IMAP backend (Reportedby David Duncan Ross Palmer)
* ASTERISK-24274 - [patch]Codec Format Is Not Included in the SDPMedia Attributes When SLIN48 Codec Is Used (Reported by FrankieChin)
* ASTERISK-24533 - 2 threads created per chan_sip entry (Reportedby xrobau)
* ASTERISK-24542 - [patch]Failure showing codecs via 'core showchanneltype
* ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLsallow blocked addresses through (Reported by Matt Jordan)
* ASTERISK-24534 - [patch]Register DB() as escalating to preventusers from writing to astdb (Reported by Gareth Palmer)
* ASTERISK-24531 - res_pjsip_acl: ACLs not applied on initialmodule load (Reported by Matt Jordan)
* ASTERISK-24490 - Security Vulnerability: CONFBRIDGE function'srecord_command option allows arbitrary parameters to be passedto MixMonitor, allowing remote execution of commands (Reportedby Matt Jordan)
* ASTERISK-24528 - res_pjsip_refer: Sending INVITE with Replacesin-dialog with invalid target causes crash (Reported by JoshuaColp)
* ASTERISK-24471 - Crash - assert_fail in libc inpjmedia_sdp_neg_negotiate from /usr/local/lib/libpjmedia.so.2(Reported by yaron nahum)
* ASTERISK-24535 - stringfields: Fix regression from fix forunintentional memory retention and another issue exposed by thefix (Reported by Corey Farrell)
* ASTERISK-24508 - pjsip - REFER request from SNOM is rejectedwith "400 bad request" - DEBUG shows "Received a REFER without aparseable 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 allendmarked users when last marked user leaves (Reported by MattJordan)
* ASTERISK-23651 - Reloading some modules that are loaded already,results in 'No such module' before a successful reload (Reportedby 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 followedby a hangup can cause an ARI client to not receive an expectedChannelLeftBridge event before StasisEnd (Reported by MattJordan)
* ASTERISK-24489 - Crash: Asterisk crashes when converting RTCPpacket to JSON for res_hep_rtcp and report blocks are greaterthan 1 (Reported by Gregory Malsack)
* ASTERISK-24498 - Segmentation fault in res_hep_rtcp on attendedtransfer (Reported by Beppo Mazzucato)
* ASTERISK-24281 - When bridging 2 chan_sip channels, MOH notremoved from on-hold channels and bridge is never destroyedafter hangup. (Reported by Stefan Engström)
* ASTERISK-24444 - PBX: Crash when generating extension forpattern matching hint (Reported by Leandro Dardini)
* ASTERISK-24502 - Build fails when dev-mode, dont optimize andcoverage 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 SVNrevision r227276 (Reported by Xavier Hienne)
* ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMSlength exceeds 50 (roughly) national symbols (Reported byDmitriy 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 whenwaiting for more matching digits. (Reported by Richard Mudgett)
* ASTERISK-24257 - agent must dial acceptdtmf twice to bridge toqueue caller (Reported by Steve Pitts)
* ASTERISK-24492 - main/file.c: ast_filestream sometimes causesextra calls to ast_module_unref (Reported by Corey Farrell)
* ASTERISK-24491 - Memory leak in res_hep (Reported by ZaneConkle)
* ASTERISK-24307 - Unintentional memory retention in stringfields(Reported by Etienne Lessard)
* ASTERISK-24438 - res_pjsip_multihomed.so blocks Asterisk reloadwhen 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 GeorgeJoseph)
* ASTERISK-24487 - configuration: sections should be loadable astemplate even when not marked (Reported by Scott Griepentrog)
* ASTERISK-24482 - func_talkdetect: Fix stasis message leak inaudiohook callback (Reported by Corey Farrell)
* ASTERISK-24480 - res_http_websockets: Module reference decreasebelow zero (Reported by Corey Farrell)
* ASTERISK-24476 - main/app.c / app_voicemail: ast_writestreamleaks (Reported by Corey Farrell)
* ASTERISK-24411 - [patch] Status of outbound registration is notchanged 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 structcall_queue (Reported by Corey Farrell)
* ASTERISK-24465 - audiohooks list leaks reference to formats(Reported by Corey Farrell)
* ASTERISK-24462 - res_pjsip: Stale qualify statistics afterdisablementation (Reported by Kevin Harwell)
* ASTERISK-24190 - IMAP voicemail causes segfault (Reported byNick Adams)
* ASTERISK-24304 - asterisk crashing randomly because of unistimchannel (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 byCorey Farrell)
* ASTERISK-24453 - manager: acl_change_sub leaks (Reported byCorey Farrell)
* ASTERISK-24437 - Review implementation of ast_bridge_impart forleaks and document proper usage (Reported by Scott Griepentrog)
* ASTERISK-24430 - missing letter "p" in word response inOriginateResponse 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 inconfigs/extensions.conf.sample (Reported by Ben Klang)
* ASTERISK-24454 - app_queue: ao2_iterator not destroyed, causingleak (Reported by Corey Farrell)
* ASTERISK-24455 - func_cdr: CDR_PROP leaks payload (Reported byCorey Farrell)
* ASTERISK-24435 - Asterisk 13 with TC400P segfault (Reported byMarian Koniuszko)
* ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead ofSSLv3, security fix POODLE (CVE-2014-3566) (Reported byabelbeck)
* ASTERISK-24122 - Documentaton for res_pjsip option use_avpfneeds to be fixed (Reported by James Van Vleet)
* ASTERISK-24381 - res_pjsip_sdp_rtp: Declined media streams areinterpreted, leading to erroneous 488 rejections (Reported byMatt Jordan)
* ASTERISK-24063 - [patch]Asterisk does not respect outbound proxywhen sending qualify requests (Reported by Damian Ivereigh)
* ASTERISK-24415 - Missing AMI VarSet events when channels inheritvariables. (Reported by Richard Mudgett)
* ASTERISK-24327 - bridge_native_rtp: Smart bridge operation tosoftmix sometimes fails to properly re-INVITE remotely bridgedparticipants (Reported by Matt Jordan)
* ASTERISK-24426 - CDR Batch mode: size used as time value afterfirst expire (Reported by Shane Blaser)
* ASTERISK-24312 - SIGABRT when improperly configured realtimepjsip (Reported by Dafi Ni)
* ASTERISK-23846 - Unistim multilines. Loss of voice after secondcall drops (on a second line). (Reported by Rustam Khankishyiev)
* ASTERISK-24413 - parking/parking_tests: Crash due to assertionin 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 queuestests (Reported by Steve Pitts)
* ASTERISK-24392 - res_fax: fax gateway sessions leak (Reported byCorey 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 byTzafrir 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 differentlysince 11.13.0 (Reported by Etienne Lessard)
* ASTERISK-24387 - res_pjsip: rport sent from UAS MUST include theport that the UAC sent the request on (Reported by Matt Jordan)
* ASTERISK-20784 - Failure to receive an ACK to a SIP Re-INVITEresults in a SIP channel leak (Reported by NITESH BANSAL)
* ASTERISK-15879 - [patch] Failure to receive an ACK to a SIPRe-INVITE results in a SIP channel leak (Reported by TorreySearle)
* ASTERISK-24383 - res_rtp_asterisk: Crash if no candidatesreceived for component (Reported by Kevin Harwell)
* ASTERISK-24011 - [patch]safe_asterisk tries to set ulimit -n toohigh on linux systems with lots of RAM (Reported by MichaelMyles)
* ASTERISK-24326 - res_rtp_asterisk: ICE-TCP candidates areincorrectly 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 clientstate to the configuration parameter value (Reported by MattJordan)
* 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 sentto Asterisk with no user in request is always 404'd (Reported byMatt Jordan)
* ASTERISK-24382 - chan_pjsip: Calling PJSIP_MEDIA_OFFER on anon-PJSIP channel results in an invalid reference of a channelpvt and a FRACK (Reported by Matt Jordan)
* ASTERISK-24369 - res_pjsip: Large message on reliable transportcan 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 persistencecauses crash when re-constructing stored subscription (Reportedby Matt Jordan)
* ASTERISK-24378 - Release AMI connections on shutdown (Reportedby Corey Farrell)
* ASTERISK-24384 - chan_motif: format capabilities leak on moduleload error (Reported by Corey Farrell)
* ASTERISK-24199 - 'ALL' is specified in pjsip.conf.sample for TLScipher but it is not valid (Reported by Joshua Colp)
* ASTERISK-24195 - bridge_native_rtp: Removing mixmonitor from anative RTP capable smart bridge doesn't cause the bridge toresume 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 outputfields and event documentation (Reported by Mitch Claborn)
* ASTERISK-23781 - outgoing missing as enum fromcontrib/ast-db-manage/config (Reported by Stephen More)
* ASTERISK-24222 - PJSIP: Failed assertions when placing a callwith 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 withrealtime peers (Reported by ibercom)
* ASTERISK-24350 - PJSIP shows commands prints unneeded headers(Reported by snuffy)
* ASTERISK-20567 - bashism in autosupport (Reported by TzafrirCohen)
* ASTERISK-24357 - [fax] Out of bounds error in update_modem_bits(Reported by Jeremy Lainé)
* ASTERISK-24348 - Built-in editline tab complete segfault withMALLOC_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 requestcrashes (Reported by Rogger Padilla)
* ASTERISK-24335 - [PATCH] Asterisk incorrectly responds 503 toINVITE retransmissions of rejected calls (Reported by TorreySearle)
* ASTERISK-24339 - Swagger API Docs have incorrect basePath(Reported by Bradley Watkins)
* ASTERISK-24265 - segfault in asterisk when try to make call toIAX (Reported by Dafi Ni)
* ASTERISK-24290 - Endpoint identifier match value fails to parsewhen CIDR network format is specified (Reported by Ray Crumrine)
* ASTERISK-24301 - Security: Out of call MESSAGE requestsprocessed via Message channel driver can crash Asterisk(Reported by Matt Jordan)
* ASTERISK-24136 - Security: Crash in Asterisk's PJSIP code whensubscribing to an event with an unexpected body type (Reportedby Mark Michelson)
* ASTERISK-24161 - PJSIPShowEndpoint gives inaccurate count oflist items (Reported by Mark Michelson)
* ASTERISK-24331 - Unexpected Errors in Asterisk Manager InterfaceOutput (Reported by xrobau)
* ASTERISK-24328 - Use of MixMonitor 'm' option results in 0duration vm description file (Reported by Scott Griepentrog)
* ASTERISK-23577 - res_rtp_asterisk: Crash inast_rtp_on_turn_rtp_state when RTP instance is NULL (Reported byJay Jideliov)
* ASTERISK-23634 - With TURN Asterisk crashes on multiple (7-10)concurrent WebRTC (avpg/encryption/icesupport) calls (Reportedby Roman Skvirsky)
* ASTERISK-24249 - SIP debugs do not stop (Reported by AvinashMohod)
* ASTERISK-24181 - RLS: Large lists don't get sent because theyexceed the PJSIP message length limit (Reported by JonathanRose)
* ASTERISK-24254 - CDRs: Application/args/dialplan CEP updatedduring dial operation (Reported by Matt Jordan)
* ASTERISK-24241 - crash: CDRs recursively attempt to update PartyB information in a multi-party bridge, overrunning the stack(Reported by Deepak Singh Rawat)
* ASTERISK-24208 - Channels with CDR Information Remain ActiveEven After ConfBrige Is Ended (Reported by Frankie Chin)
* ASTERISK-24223 - Gibberish Call-ID on Local channel onorigination (Reported by Mark Michelson)
* ASTERISK-24271 - Unable to make WebRTC call through chan_PJSIPnor chan_SIP (Reported by Dafi Ni)
* ASTERISK-24212 - testsuite: Sporadic crash due to assert onstopping RTP engine (Reported by Matt Jordan)
* ASTERISK-24264 - ARI: Adding a channel to a holding bridgeautomatically starts MOH (Reported by Samuel Galarneau)
* ASTERISK-23767 - [patch] Dynamic IAX2 registration stops tryingif 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 restartsat beginning of file. (Reported by Jason Richards)
* ASTERISK-24143 - pjsip: Outbound call to WebRTC UA fails totransmit ACK on received 200 OK (Reported by Aleksei Kulakov)
* ASTERISK-23997 - chan_sip: port incorrectly incremented for RTCPICE 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 notbe fully qualified domainname (Reported by Private Name)
* ASTERISK-22252 - res_musiconhold cleanup - REF_DEBUG reloadwarnings 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 answerschannel, preventing early media playback (Reported by MattJordan)
* ASTERISK-24245 - gcc 4.1.2 complains of files that do not endwith newlines (Reported by Shaun Ruffell)
* ASTERISK-24246 - Quiet warning about type qualifiers ignored onfunction return type (Reported by Shaun Ruffell)
* ASTERISK-24043 - ARI /continue fails to actually continue intothe dialplan (Reported by Krandon Bruse)
* ASTERISK-24215 - testsuite: ARI Live Dangerously test fails dueto wrong response code from Asterisk (Reported by Matt Jordan)
* ASTERISK-24134 - ARI: GET /channels/{channel_id}/variable forchannel in dialplan returns 409 conflict (Reported by MattJordan)
* ASTERISK-24138 - dial: Call forwarding information presentedthrough AMI/ARI is wrong (Reported by Matt Jordan)
* ASTERISK-24234 - app_meetme: Crash on conference shutdown due toNULL channel passed to meetme_stasis_generate_msg() (Reported byShaun Ruffell)
* ASTERISK-24225 - Dial option z is broken (Reported bydimitripietro)
* ASTERISK-24032 - Gentoo compilation emits warning:"_FORTIFY_SOURCE" redefined (Reported by Kilburn)
* ASTERISK-24027 - MixMonitor AMI action called during AGIexecution from bridge feature causes channel to leave AGI hashung up (Reported by Matt Jordan)
* ASTERISK-24236 - res_hep_rtcp: Module incorrectly depends onpjsip (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 MattJordan)
* ASTERISK-25471 - [patch]Add subscribe_context to res_pjsip(Reported by JoshE)
* ASTERISK-26159 - res_hep: enabled by default and informationsent to default address (Reported by Ross Beer)
* ASTERISK-26088 - Investigate heavy memory utilization byres_pjsip_pubsub (Reported by Richard Mudgett)
* ASTERISK-25578 - [patch] SIP/SDP: No rtpmap for static RTPpayload 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 multipleclients 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. (Reportedby Alexei Gradinari)
* ASTERISK-25835 - Authentication using 'Username' field fromDigest (Reported by Ross Beer)
* ASTERISK-25930 - PJSIP: disable multi domain to improve realtimeperformace (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 forAnonymous
* ASTERISK-25767 - [patch] Add check to configure for sanitizes (Reported by Badalian Vyacheslav)
* ASTERISK-25068 - Move commonly used FreePBX extra sounds to thecore set (Reported by Rusty Newton)
* ASTERISK-25627 - Easily Preventable Compile Warning (Reported byDiederik de Groot)
* ASTERISK-25558 - [patch]chan_sip option 'notifyringing' doc fixand addition of 'notifyringingprio' (Reported by Ward vanWanrooij)
* ASTERISK-25618 - res_pjsip: Check for readability of TLS filesat 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 Asteriskendpoints (Reported by Matt Jordan)
* ASTERISK-25571 - PJSIP: Add StatsD stats for some common PJSIPobjects (Reported by Matt Jordan)
* ASTERISK-25518 - taskprocessor: Add high water mark (Reported byJonathan Rose)
* ASTERISK-25495 - [patch] Prevent old-update packages onrepository Debian systems (Reported by Rodrigo RamirezNorambuena)
* ASTERISK-25477 - pjsip show "command" like [criteria] (Reportedby Bryant Zimmerman)
* ASTERISK-24718 - [patch]Add inital support of "sanitize" toconfigure (Reported by Badalian Vyacheslav)
* ASTERISK-24870 - ARI: Subscriptions to bridges generally notsuper 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 eventswhen Asterisk deletes a dialplan variable. (Reported by RichardMudgett)
* ASTERISK-25040 - pbx: Improve performance of reloads by makinghint 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_pjsipcontact lifecycle changes (Reported by George Joseph)
* ASTERISK-25072 - res_pjsip_outbound_registration: linefunctionality. 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 namesupport to Asterisk TLS support (Reported by Maciej Szmigiero)
* ASTERISK-25044 - sorcery: Add ability to insert a new wizardinto an object type's list (Reported by George Joseph)
* ASTERISK-24892 - Super Awesome Company sound prompts (Reportedby Rusty Newton)
* ASTERISK-24744 - Swedish Core Voice prompts (Reported by ToveHjelm)
* ASTERISK-25049 - CLI: Enable automatic references to modules(Reported by Corey Farrell)
* ASTERISK-25056 - Modules: Make ast_module_info->self availableto 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 (Reportedby yaron nahum)
* ASTERISK-24917 - [patch] clang compilation warnings (Reported byDiederik de Groot)
* ASTERISK-25051 - Remove unneeded uses of optional_api providers.(Reported by Corey Farrell)
* ASTERISK-24974 - Astobj2: Allow reference debugging to beenabled/disabled by config. (Reported by Corey Farrell)
* ASTERISK-24980 - cdr_adaptive_odbc: refactor lines toconcatenate of columns name (Reported by Rodrigo RamirezNorambuena)
* ASTERISK-24947 - res_pjsip: Add a PJSIP resolver using core DNS(Reported by Joshua Colp)
* ASTERISK-24965 - cel_pgsql - log_error string references CDRinstead of CEL (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-24960 - Build System: Create MOD_ADD_SOURCE macro formodule Makefiles (Reported by Corey Farrell)
* ASTERISK-24939 - [patch]IAX make calltoken expiration timeconfigurable (Reported by Y Ateya)
* ASTERISK-24918 - pjsip: add CLI options to display global andsystem configuration (Reported by Scott Griepentrog)
* ASTERISK-24862 - [patch] Support in-dialog OPTIONS (Reported byyaron nahum)
* ASTERISK-24802 - stasis: set a channel variable on websocketdisconnect error (Reported by Kevin Harwell)
* ASTERISK-24133 - [patch]Please support Clang; Allow no-execstacks (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 notuse 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 servername for security purposes (Reported by Andrew Nagy)
* ASTERISK-24671 - Missing docs for the CDR AMI Event (Reported byDan Jenkins)
* ASTERISK-24575 - [patch]Make capath work for res_pjsip (Reportedby cloos)
* ASTERISK-24678 - [PATCH] Added atxfer* settings tofeatures.conf.sample (Reported by Niklas Larsson)
* ASTERISK-24412 - [patch]Incomplete channel originate/continuehandling with ARI (Reported by Nir Simionovich (GreenfieldTech -Israel))
* ASTERISK-24351 - [patch] Allow passing options and command toMixMonitor when recording in ConfBridge (Reported by GarethPalmer)
* ASTERISK-24553 - ARI/AMI: Include language in standard channelsnapshot output (Reported by Matt Jordan)
* ASTERISK-24552 - ARI: Allow associating a channel as aninitiator of an Origination for record keeping purposes(Reported by Matt Jordan)
* ASTERISK-24577 - Speed up loopback switches by avoiding unneededlookups (Reported by Birger "WIMPy" Harzenetter)
* ASTERISK-24530 - [patch] app_record stripping 1/4 second fromrecordings (Reported by Ben Smithurst)
* ASTERISK-24283 - [patch]Microseconds precision in the eventtimecolumn in the cel_odbc module (Reported by Etienne Lessard)
* ASTERISK-24128 - [Patch] Adding default dtls settings (Reportedby Michael K.)
* ASTERISK-24279 - Documentation: Clarify the behaviour of the CDRproperty '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/headcaller channel on queue (Reported by Kristian Høgh)
* ASTERISK-23324 - [patch] - QLOOG commiting Japanese translatedprompts (Reported by Kevin McCoy)
* ASTERISK-24038 - device state: Report ONHOLD device state ifchannel driver defers device state calculation to core (Reportedby Matt Jordan)
* ASTERISK-24171 - [patch] Provide a manpage for the aelparseutility (Reported by Jeremy Lainé)
* ASTERISK-23953 - Testsuite: Off-nominal Authenticate test(Reported by Matt Jordan)
* ASTERISK-24045 - [patch]Voicemail to email at multiple emailaddresses (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!