Rilasciato Asterisk 13.8-cert1
Il giorno 13 luglio 2016, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk Asterisk 13.8-cert1.
Dal post originale:
The release of Certified Asterisk 13.8-cert1 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 release:
New Features made in this release:
-----------------------------------
* ASTERISK-24919 - res_pjsip_config_wizard: Ability to write contents to file (Reported by Ray Crumrine)
* ASTERISK-25670 - Add regcontext to PJSIP (Reported by Daniel Journo)
* ASTERISK-25480 - [patch]Add field PauseReason on QueueMemberStatus (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25419 - Dialplan Application for Integration of StatsD (Reported by Ashley Sanders)
* ASTERISK-25549 - Confbridge: Add participant timeout option (Reported by Mark Michelson)
* ASTERISK-24922 - ARI: Add the ability to intercept hold and raise an event (Reported by Matt Jordan)
* ASTERISK-25377 - res_pjsip: Change default "From user" from UUID to something more palatable (Reported by Mark Michelson)
* ASTERISK-25252 - ARI: Add the ability to manipulate log channels (Reported by Matt Jordan)
* ASTERISK-25259 - chan_pjsip: Add rtptimeout support (Reported by Joshua Colp)
* ASTERISK-25238 - ARI: Support push configuration (Reported by Matt Jordan)
* ASTERISK-25173 - ARI: Add the ability to load/reload/unload an Asterisk module (Reported by Matt Jordan)
* ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation (Reported by Dwayne Hubbard)
* ASTERISK-24703 - ARI: Add the ability to "transfer" (redirect) a channel (Reported by Matt Jordan)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua Colp)
* ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported by George Joseph)
* ASTERISK-26099 - res_pjsip_pubsub: Crash when sending request due to server timeout (Reported by Ross Beer)
* ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George Joseph)
* ASTERISK-26089 - Invalid security events during boot using PJSIP Realtime (Reported by Scott Griepentrog)
* ASTERISK-25885 - res_pjsip: Race condition between adding contact and automatic expiration (Reported by Joshua Colp)
* ASTERISK-26074 - res_odbc: Deadlock within UnixODBC (Reported by Ross Beer)
* ASTERISK-26054 - Asterisk crashes (core dump) (Reported by B. Davis)
* ASTERISK-26034 - T.38 passthrough problem behind firewall due to early nosignal packet (Reported by George Joseph)
* ASTERISK-26030 - call cut because of double Session-Expires header in re-invite after proxy authentication is required (Reported by George Joseph)
* ASTERISK-26004 - res_pjsip: The transport/method parameter is ignored (Reported by George Joseph)
* ASTERISK-25978 - res_pjsip_authenticator_digest: Should not use source port in nonce verification (Reported by Mark Michelson)
* ASTERISK-25998 - file: Crash when using nativeformats (Reported by Joshua Colp)
* ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive sometimes two calls (Reported by nik600)
* ASTERISK-25951 - res_agi: run_agi eats frames it shouldn't (Reported by George Joseph)
* ASTERISK-25947 - Protocol transfers to stasis applications are missing the StasisStart with the replace_channel object. (Reported by Richard Mudgett)
* ASTERISK-24649 - Pushing of channel into bridge fails; Stasis fails to get app name (Reported by John Bigelow)
* ASTERISK-24782 - StasisEnd event not present for channel that was swapped out for another after completing attended transfer (Reported by John Bigelow)
* ASTERISK-25942 - res_pjsip_caller_id: Transfer results in mixed ConnectedLine information (Reported by George Joseph)
* ASTERISK-25928 - res_pjsip: URI validation done outside of PJSIP thread (Reported by Joshua Colp)
* ASTERISK-25929 - res_pjsip_registrar: AOR_CONTACT_ADDED events not raised (Reported by Joshua Colp)
* ASTERISK-25796 - res_pjsip: DOS/Crash when TCP/TLS sockets exceed pjproject PJ_IOQUEUE_MAX_HANDLES (Reported by George Joseph)
* ASTERISK-25707 - Long contact URIs or hostnames can crash pjproject/Asterisk under certain conditions (Reported by George Joseph)
* ASTERISK-25854 - No audio after HOLD/RESUME - incorrect a=recvonly in SDP from Asterisk (Reported by Robert McGilvray)
* ASTERISK-25882 - ARI: Crash can occur due to race condition when attempting to operate on a hung up channel (Part 2) (Reported by Richard Mudgett)
* ASTERISK-25849 - chan_pjsip: transfers with direct media sometimes drops audio (Reported by Kevin Harwell)
* ASTERISK-25113 - install_prereq in Debian 8 without "standard system utilities" (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25814 - Segfault at f ip in res_pjsip_refer.so (Reported by Sergio Medina Toledo)
* ASTERISK-25023 - Deadlock in chan_sip in update_provisional_keepalive (Reported by Arnd Schmitter)
* ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local channel (Reported by Filip Frank)
* ASTERISK-25829 - res_pjsip: PJSIP does not accept spaces when separating multiple AORs (Reported by Mateusz Kowalski)
* ASTERISK-25771 - ARI:Crash - Attended transfers of channels into Stasis application. (Reported by Javier Riveros )
* ASTERISK-25830 - Revision 2451d4e breaks NAT (Reported by Sean Bright)
* ASTERISK-25582 - Testsuite: Reactor timeout error in tests/fax/pjsip/directmedia_reinvite_t38 (Reported by Matt Jordan)
* ASTERISK-25811 - Unable to delete object from sorcery cache (Reported by Ross Beer)
* ASTERISK-25800 - [patch] Calculate talktime when is first call answered (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25727 - RPM build requires OPTIONAL_API cflag due to PJSIP requirement (Reported by Gergely Dömsödi)
* ASTERISK-25337 - Crash on PJSIP_HEADER Add P-Asserted-Identity when calling from Gosub (Reported by Jacques Peacock)
* ASTERISK-25738 - res_pjsip_pubsub: Crash while executing OutboundSubscriptionDetail ami action (Reported by Kevin Harwell)
* ASTERISK-25721 - [patch] res_phoneprov: memory leak and heap-use-after-free (Reported by Badalian Vyacheslav)
* ASTERISK-25272 - [patch]The ICONV dialplan function sometimes returns garbage (Reported by Etienne Lessard)
* ASTERISK-25751 - res_pjsip: Support pjsip_dlg_create_uas_and_inc_lock (Reported by Joshua Colp)
* ASTERISK-25606 - Core dump when using transports in sorcery (Reported by Martin MouÄka)
* ASTERISK-20987 - non-admin users, who join muted conference are not being muted (Reported by hristo)
* ASTERISK-25737 - res_pjsip_outbound_registration: line option not in Alembic (Reported by Joshua Colp)
* ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in udptl_rx_packet cause ast_frdup crash (Reported by Walter Doekes)
* ASTERISK-25742 - Secondary IFP Packets can result in accessing uninitialized pointers and a crash (Reported by Torrey Searle)
* ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST Vulnerability - Investigate vulnerability of HTTP server (Reported by Alex A. Welzl)
* ASTERISK-25397 - [patch]chan_sip: File descriptor leak with non-default timert1 (Reported by Alexander Traud)
* ASTERISK-25702 - PjSip realtime DB and Cache Errors since upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported by Nic Colledge)
* ASTERISK-25730 - build: make uninstall after make distclean tries to remove root (Reported by George Joseph)
* ASTERISK-25725 - core: Incorrect XML documentation may result in weird behavior (Reported by Joshua Colp)
* ASTERISK-25722 - ASAN & testsute: stack-buffer-overflow in sip_sipredirect (Reported by Badalian Vyacheslav)
* ASTERISK-25709 - ARI: Crash can occur due to race condition when attempting to operate on a hung up channel (Reported by Mark Michelson)
* ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported by Badalian Vyacheslav)
* ASTERISK-25685 - infrastructure: Run alembic in Jenkins build script (Reported by Joshua Colp)
* ASTERISK-25712 - Second call to already-on-call phone and Asterisk sends "Ready" (Reported by Richard Mudgett)
* ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow (Reported by Badalian Vyacheslav)
* ASTERISK-25179 - CDR(billsec,f) and CDR(duration,f) report incorrect values (Reported by Gianluca Merlo)
* ASTERISK-25611 - core: threadpool thread_timeout_thrash unit test sporadically failing (Reported by Joshua Colp)
* ASTERISK-24097 - Documentation - CHANNEL function help text missing 'linkedid' argument (Reported by Steven T. Wheeler)
* ASTERISK-25700 - main/config: Clean config maps on shutdown. (Reported by Corey Farrell)
* ASTERISK-25696 - bridge_basic: don't cache xferfailsound during a transfer (Reported by Kevin Harwell)
* ASTERISK-25697 - bridge_basic: don't play an attended transfer fail sound after target hangs up (Reported by Kevin Harwell)
* ASTERISK-25683 - res_ari: Asterisk fails to start if compiled with MALLOC_DEBUG (Reported by yaron nahum)
* ASTERISK-25686 - PJSIP: qualify_timeout is a double, database schema is an integer (Reported by Marcelo Terres)
* ASTERISK-25690 - Hanging up when executing connected line sub does not cause hangup (Reported by Joshua Colp)
* ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh reload' cause a crash (Reported by Sean Bright)
* ASTERISK-25632 - res_pjsip_sdp_rtp: RTP is sent from wrong IP address when multihomed (Reported by Olivier Krief)
* ASTERISK-25637 - Multi homed server using wrong IP (Reported by Daniel Journo)
* ASTERISK-25394 - pbx: Incorrect device and presence state when changing hint details (Reported by Joshua Colp)
* ASTERISK-25640 - pbx: Deadlock on features reload and state change hint. (Reported by Krzysztof Trempala)
* ASTERISK-25681 - devicestate: Engine thread is not shut down (Reported by Corey Farrell)
* ASTERISK-25680 - manager: manager_channelvars is not cleaned at shutdown (Reported by Corey Farrell)
* ASTERISK-25679 - res_calendar leaks scheduler. (Reported by Corey Farrell)
* ASTERISK-25675 - Endpoint not listed as Unreachable (Reported by Daniel Journo)
* ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported by Corey Farrell)
* ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey Farrell)
* ASTERISK-25668 - res_pjsip: Deadlock in distributor (Reported by Mark Michelson)
* ASTERISK-25664 - ast_format_cap_append_by_type leaks a reference (Reported by Corey Farrell)
* ASTERISK-25647 - bug of cel_radius.c: wrong point of ADD_VENDOR_CODE (Reported by Aaron An)
* ASTERISK-25317 - asterisk sends too many stun requests (Reported by Stefan Engström)
* ASTERISK-25137 - endpoint stasis messages are delivered twice (Reported by Vitezslav Novy)
* ASTERISK-25116 - res_pjsip: Two PeerStatus AMI messages are sent for every status change (Reported by George Joseph)
* ASTERISK-25641 - bridge: GOTO_ON_BLINDXFR doesn't work on transfer initiated channel (Reported by Dmitry Melekhov)
* ASTERISK-25614 - DTLS negotiation delays (Reported by Dade Brandon)
* ASTERISK-25442 - using realtime (mysql) queue members are never updated in wait_our_turn function (app_queue.c) (Reported by Carlos Oliva)
* ASTERISK-25625 - res_sorcery_memory_cache: Add full backend caching (Reported by Joshua Colp)
* ASTERISK-25601 - json: Audit reference usage and thread safety (Reported by Joshua Colp)
* ASTERISK-25615 - res_pjsip: Setting transport async_operations > 1 causes segfault on tls transports (Reported by George Joseph)
* ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and thread of asterisk is not released (Reported by Hiroaki Komatsu)
* ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by sungtae kim)
* ASTERISK-25619 - res_chan_stats not sending the correct information to StatsD (Reported by Tyler Cambron)
* ASTERISK-25569 - app_meetme: Audio quality issues (Reported by Corey Farrell)
* ASTERISK-25609 - [patch]Asterisk may crash when calling ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
* ASTERISK-24146 - [patch]No audio on WebRtc caller side when answer waiting time is more than ~7sec (Reported by Aleksei Kulakov)
* ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec (Reported by Alexander Traud)
* ASTERISK-25616 - Warning with a Codec Module which supports PLC with FEC (Reported by Alexander Traud)
* ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by Dudás József)
* ASTERISK-25608 - res_pjsip/contacts/statsd: Lifecycle events aren't consistent (Reported by George Joseph)
* ASTERISK-25584 - [patch] format-attribute module: VP8 missing (Reported by Alexander Traud)
* ASTERISK-25583 - [patch] format-attribute module: RFC 7587 (Opus Codec) (Reported by Alexander Traud)
* ASTERISK-25498 - Asterisk crashes when negotiating g729 without that module installed (Reported by Ben Langfeld)
* ASTERISK-25595 - Unescaped : in messge sent to statsd (Reported by Niklas Larsson)
* ASTERISK-25476 - chan_sip loses registrations after a while (Reported by Michael Keuter)
* ASTERISK-25598 - res_pjsip: Contact status messages are printing a hash instead of the uri (Reported by George Joseph)
* ASTERISK-25600 - bridging: Inconsistency in BRIDGEPEER (Reported by Jonathan Rose)
* ASTERISK-25593 - fastagi: record file closed after sending result (Reported by Kevin Harwell)
* ASTERISK-25585 - [patch]rasterisk never hits most of main(), but it's assumed to (Reported by Walter Doekes)
* ASTERISK-25590 - CLI Usage info for 'pjsip send notify' references incorrect config (Reported by Corey Farrell)
* ASTERISK-25165 - Testsuite - Sorcery memory cache leaks (Reported by Corey Farrell)
* ASTERISK-25575 - res_pjsip: Dynamic outbound registrations created via ARI are not loaded into memory on Asterisk start/restart (Reported by Matt Jordan)
* ASTERISK-25545 - [patch] translation module gets cached not joint format (Reported by Alexander Traud)
* ASTERISK-25573 - [patch] H.264 format attribute module: resets whole SDP (Reported by Alexander Traud)
* ASTERISK-24958 - Forwarding loop detection inhibits certain desirable scenarios (Reported by Mark Michelson)
* ASTERISK-25561 - app_queue.c line 6503 (try_calling): mutex 'qe->chan' freed more times than we've locked! (Reported by Alec Davis)
* ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by Joshua Colp)
* ASTERISK-25160 - [patch] Opus Codec: SIP/SDP line fmtp missing when called internally (Reported by Alexander Traud)
* ASTERISK-25535 - [patch] format creation on module load instead of cache (Reported by Alexander Traud)
* ASTERISK-25449 - main/sched: Regression introduced by 5c713fdf18f causes erroneous duplicate RTCP messages; other potential scheduling issues in chan_sip/chan_skinny (Reported by Matt Jordan)
* ASTERISK-25546 - threadpool: Race condition between idle timeout and activation (Reported by Joshua Colp)
* ASTERISK-25537 - [patch] format-attribute module: RFC or internal defaults? (Reported by Alexander Traud)
* ASTERISK-25533 - [patch] buffer for ast_format_cap_get_names only 64 bytes (Reported by Alexander Traud)
* ASTERISK-25373 - add documentation for CALLERID(pres) and also the CONNECTEDLINE and REDIRECTING variants (Reported by Walter Doekes)
* ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by Walter Doekes)
* ASTERISK-24779 - Passthrough OPUS codec not working with chan_pjsip (Reported by PowerPBX)
* ASTERISK-25522 - ARI: Crash when creating channel via ARI originate with requesting channel (Reported by Matt Jordan)
* ASTERISK-25434 - Compiler flags not reported in 'core show settings' despite usage during compilation (Reported by Rusty Newton)
* ASTERISK-24106 - WebSockets Automatically decides what driver it will use (Reported by Andrew Nagy)
* ASTERISK-25513 - Crash: malloc failed with high load of subscriptions. (Reported by John Bigelow)
* ASTERISK-25505 - res_pjsip_pubsub: Crash on off-nominal when UAS dialog can't be created (Reported by Joshua Colp)
* ASTERISK-24543 - Asterisk 13 responds to SIP Invite with all possible codecs configured for peer as opposed to intersection of configured codecs and offered codecs (Reported by Taylor Hawkes)
* ASTERISK-25494 - build: GCC 5.1.x catches some new const, array bounds and missing paren issues (Reported by George Joseph)
* ASTERISK-25485 - res_pjsip_outbound_registration: registration stops due to 400 response (Reported by Kevin Harwell)
* ASTERISK-25486 - res_pjsip: Fix deadlock when validating URIs (Reported by Joshua Colp)
* ASTERISK-7803 - [patch] Update the maximum packetization values in frame.c (Reported by dea)
* ASTERISK-25484 - [patch] autoframing=yes has no effect (Reported by Alexander Traud)
* ASTERISK-25461 - Nested dialplan #includes don't work as expected. (Reported by Richard Mudgett)
* ASTERISK-25455 - Deadlock of PJSIP realtime over res_config_pgsql (Reported by mdu113)
* ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing (Reported by Olle Johansson)
* ASTERISK-25435 - Asterisk periodically hangs. UDP Recv-Q greatly exceeds zero. (Reported by Dmitriy Serov)
* ASTERISK-25451 - Broken video - erased rtp marker bit (Reported by Stefan Engström)
* ASTERISK-25400 - Hints broken when "CustomPresence" doesn't exist in AstDB (Reported by Andrew Nagy)
* ASTERISK-25443 - [patch]IPv6 - Potential issue in via header parsing (Reported by ffs)
* ASTERISK-25404 - segfault/crash in chan_pjsip_hangup ... at chan_pjsip.c (Reported by Chet Stevens)
* ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON (Reported by Bojan NemÄić)
* ASTERISK-25441 - Deadlock in res_sorcery_memory_cache. (Reported by Richard Mudgett)
* ASTERISK-25438 - res_rtp_asterisk: ICE role message even when ICE is not enabled (Reported by Joshua Colp)
* ASTERISK-25383 - Core dumps on startup and shutdown with MALLOC_DEBUG enabled (Reported by yaron nahum)
* ASTERISK-25423 - Caller gets no Connected line update during call pickup. (Reported by Richard Mudgett)
* ASTERISK-25305 - Dynamic logger channels can be added multiple times (Reported by Mark Michelson)
* ASTERISK-25418 - On-hold channels redirected out of a bridge appear to still be on hold (Reported by Mark Michelson)
* ASTERISK-25384 - Regular Asterisk crashes when using Page application. "user_data is NULL" (Reported by Chet Stevens)
* ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations (Reported by Elazar Broad)
* ASTERISK-25410 - app_record: RECORDED_FILE variable not being populated (Reported by Kevin Harwell)
* ASTERISK-25396 - chan_sip: Extremely long callerid name causes invalid SIP (Reported by Walter Doekes)
* ASTERISK-25399 - app_queue: AgentComplete event has wrong reason (Reported by Kevin Harwell)
* ASTERISK-25185 - Segfault in app_queue on transfer scenarios (Reported by Etienne Lessard)
* ASTERISK-25353 - [patch] Transcoding while different in Frame size = Frames lost (Reported by Alexander Traud)
* ASTERISK-25325 - ARI PUT reload chan_sip HTTP response 404 (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25390 - default_from_user can crash with certain configuration backends (Reported by Mark Michelson)
* ASTERISK-25387 - res_pjsip_nat: Malformed REGISTER request causes NAT'd Contact header to not be rewritten (Reported by Matt Jordan)
* ASTERISK-25227 - No audio at in-band announcements in ooh323 channel (Reported by Alexandr Dranchuk)
* ASTERISK-25369 - res_parking: ParkAndAnnounce - Inheritable variables aren't applied to the announcer channel (Reported by Jonathan Rose)
* ASTERISK-25295 - res_pjsip crash - pjsip_uri_get_uri at /usr/include/pjsip/sip_uri.h (Reported by Dmitriy Serov)
* ASTERISK-25381 - res_pjsip: AoRs deleted via ARI (or other mechanism) do not destroy their related contacts (Reported by Matt Jordan)
* ASTERISK-25352 - res_hep_rtcp correlation_id is different then res_hep (Reported by Kevin Scott Adams)
* ASTERISK-25367 - pbx: Long pattern match hints may cause "core show hints" to crash (Reported by Joshua Colp)
* ASTERISK-25365 - Persistent subscriptions have extra Content-Length/corrupted messages (Reported by Mark Michelson)
* ASTERISK-25362 - Deadlock due to presence state callback (Reported by Mark Michelson)
* ASTERISK-25356 - res_pjsip_sdp_rtp: Multiple keepalive scheduled items may exist (Reported by Joshua Colp)
* ASTERISK-25355 - sched: ast_sched_del may return prematurely due to spurious wakeup (Reported by Joshua Colp)
* ASTERISK-25318 - tests/rest_api/applications/subscribe-endpoint/nominal/resource: Sporadically failing (Reported by Joshua Colp)
* ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup cause on call pickup (Reported by Joshua Colp)
* ASTERISK-25342 - res_pjsip: Repeated usage of pj_gethostip may block (Reported by Joshua Colp)
* ASTERISK-25341 - bridge: Hangups may get lost when executing actions (Reported by Joshua Colp)
* ASTERISK-25339 - res_pjsip: Empty "auth" sections from non-config backgrounds are interpreted as valid (Reported by Matt Jordan)
* ASTERISK-25215 - Differences in queue.log between Set QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne Gaetz)
* ASTERISK-25322 - Crash occurs when using MixMonitor with t() or r() options. (Reported by Richard Mudgett)
* ASTERISK-25320 - chan_sip.c: sip_report_security_event searches for wrong or non existent peer on invite (Reported by Kevin Harwell)
* ASTERISK-25315 - DAHDI channels send shortened duration DTMF tones. (Reported by Richard Mudgett)
* ASTERISK-25312 - res_http_websocket: Terminate connection on fatal cases (Reported by Joshua Colp)
* ASTERISK-25306 - Persistent subscriptions can save multiple SIP messages at once, leading to potential crashes. (Reported by Mark Michelson)
* ASTERISK-25309 - [patch] iLBC 20 advertised (Reported by Alexander Traud)
* ASTERISK-25304 - res_pjsip: XML sanitization may write past buffer (Reported by Joshua Colp)
* ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on Firefox 39 - add ECDH support and fallback to prime256v1 (Reported by Stefan Engström)
* ASTERISK-25296 - RTP performance issue with several channel drivers. (Reported by Richard Mudgett)
* ASTERISK-25297 - Crashes running channels/pjsip/resolver/srv/failover/in_dialog testsuite tests (Reported by Richard Mudgett)
* ASTERISK-25292 - Testuite: tests/apps/bridge/bridge_wait/bridge_wait_e_options fails (Reported by Kevin Harwell)
* ASTERISK-25271 - Parking & blind transfer: Transferer channel not hung up if no MOH (Reported by Kevin Harwell)
* ASTERISK-25250 - chan_sip - Despite the channel being answered, caller on a call established via Local channel continues to hear ringback (Reported by Etienne Lessard)
* ASTERISK-25253 - confbridge volume options and other volume controls such as func_volume don't work (Reported by Dmitriy Serov)
* ASTERISK-25247 - choppy audio when spying on a g722 channel, chan_sip or chan_pjsip (Reported by hristo)
* ASTERISK-24867 - Docs for 'e' option in ResetCDR say to use CDR_PROP instead, CDR_PROP docs are unclear (Reported by Rusty Newton)
* ASTERISK-24853 - Documentation claims chan_sip outbound registrations support WS or WSS as valid transports (not true) (Reported by PSDK)
* ASTERISK-25242 - PJSIP: No audio when Asterisk inside NAT and endpoints outside NAT - implement functionality similar to chan_sip 'rtpkeepalive'? (Reported by Mark Michelson)
* ASTERISK-25258 - chan_pjsip: Incorrect format switch on received RTP packet (Reported by Joshua Colp)
* ASTERISK-25257 - [patch]channels/sig_pri.h -> sig_pri_span -> force_restart_unavailable_chans in wrong scope (Reported by Patric Marschall)
* ASTERISK-24934 - [patch]Asterisk manager output does not escape control characters (Reported by warren smith)
* ASTERISK-25255 - Missing AMI VarSet events when setting to an empty string. (Reported by Richard Mudgett)
* ASTERISK-25254 - Crash if dialplan sets ATTENDEDTRANSFER to an empty string before Park. (Reported by Richard Mudgett)
* ASTERISK-25183 - PJSIP: Crash on NULL channel in chan_pjsip_incoming_response despite previous checks for NULL channel (Reported by Matt Jordan)
* ASTERISK-25201 - Crash in PJSIP distributor on already free'd threadpool (Reported by Matt Jordan)
* ASTERISK-25240 - bridge_native_rtp: Direct media wrongfully started when completing attended transfer (Reported by Joshua Colp)
* ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes (Reported by Rusty Newton)
* ASTERISK-22805 - res_rtp_asterisk: Crash when calling BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP (Reported by Dmitry Burilov)
* ASTERISK-24550 - res_rtp_asterisk: Crash in ast_rtp_on_ice_complete during DTLS handshake (Reported by Osaulenko Alexander)
* ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by Badalian Vyacheslav)
* ASTERISK-24832 - [patch]DTLS-crashes within openssl (Reported by Stefan Engström)
* ASTERISK-25127 - DTLS crashes following "Unable to cancel schedule ID" in dtls_srtp_check_pending (Reported by Dade Brandon)
* ASTERISK-25168 - Random Core Dumps on Asterisk 13.4 PJSIP, in ast_channel_name at channel_internal_api.c (Reported by Carl Fortin)
* ASTERISK-25115 - Crash related to func sip_resolve_invoke_user_callback of res_pjsip/pjsip_resolver.c (Reported by John Bigelow)
* ASTERISK-25226 - chan_sip: Channel leak in branch 13 on early replaces call pickup (Reported by Walter Doekes)
* ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c (Reported by Walter Doekes)
* ASTERISK-25219 - [patch]Source and destination overlap in memcpy in rtp_engine.c (Reported by Walter Doekes)
* ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS (Reported by Walter Doekes)
* ASTERISK-19277 - [patch]endlessly repeating error: "poll failed: Bad file descriptor" (Reported by Barry Chern)
* ASTERISK-25202 - Hints extension state broken between 13.3.2 and 13.4 (Reported by cervajs)
* ASTERISK-25196 - res_pjsip_nat: rewrite_contact should not be applied to Contact header when Record-Route headers are present (Reported by Mark Michelson)
* ASTERISK-24907 - res_pjsip_outbound_registration: crash during unload if registration attempts are still occuring (Reported by Kevin Harwell)
* ASTERISK-25204 - res_pjsip_refer: Duplicated Referred-By or Replaces headers on outbound INVITEs. (Reported by Mark Michelson)
* ASTERISK-25171 - Early completion of feature code attended transfer results in intermittent one-way audio, "ghost ringing" and robotic sound. (Reported by Rusty Newton)
* ASTERISK-25189 - AMI: Add Linkedid header to standard channel snapshot information. (Reported by Richard Mudgett)
* ASTERISK-25172 - Crash in channels/sip/sip blind transfer/caller_refer_only test in ast_format_cap_append_from_cap during ast_request (Reported by Matt Jordan)
* ASTERISK-25180 - res_pjsip_mwi: Unsolicited MWI requires reload (Reported by Joshua Colp)
* ASTERISK-25182 - [patch] on CLI sip reload, new codecs get appended only (Reported by Alexander Traud)
* ASTERISK-25163 - Deadlock in chan_sip between reload of sip peer container and MWI Stasis callback (Reported by Dmitriy Serov)
* ASTERISK-25091 - Asterisk REST API - bridge.addChannel crash asterisk when calling channel hangup while adding to bridge (Reported by Ilya Trikoz)
* ASTERISK-24900 - Manager event ParkedCallSwap is not documented (Reported by Rusty Newton)
* ASTERISK-25162 - func_pjsip_aor: Leak of contact in iterator (Reported by Corey Farrell)
* ASTERISK-25158 - res_pjsip: Add option to use AAL2 packing when negotiating g.726 (Reported by Kevin Harwell)
* ASTERISK-24344 - CDR_PROP(disable) disables CDR only for first dialed party (Reported by Janusz Karolak)
* ASTERISK-24443 - CDR fields (dst, dcontext) empty in transfer call started from Macro (Reported by Arveno Santoro)
* ASTERISK-25154 - [patch]fromtag may need to be updated after successful call dialog match (Reported by Damian Ivereigh)
* ASTERISK-25156 - chan_pjsip’s CHAN_START cel event lacks the correct context and exten (Reported by cloos)
* ASTERISK-25157 - bridging: Performing a blonde transfer does not result in connected line updates (Reported by Joshua Colp)
* ASTERISK-25087 - Asterisk segfault when using Directory application with alias option and specific mailbox configuration (Reported by Chet Stevens)
* ASTERISK-24983 - IAX deadlock between hangup and scheduled actions (ex. largrq) (Reported by Y Ateya)
* ASTERISK-25096 - [patch]Segfault when registering over websockets with PJSIP (in ast_sockaddr_isnull at /include/asterisk/netsock2.h) (Reported by Josh Kitchens)
* ASTERISK-24963 - ASAN: heap-use-after-free with PJSIP and WSS (Reported by Badalian Vyacheslav)
* ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute but asterisk doesn't detect it. (Reported by ibercom)
* ASTERISK-25094 - PBX core: Investigate thread safety issues (Reported by Corey Farrell)
* ASTERISK-25148 - res_pjsip NULL channel audit (Reported by Mark Michelson)
* ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm | adpcm | ipc10} (Reported by Badalian Vyacheslav)
* ASTERISK-25131 - chan_pjsip: In-dialog authentication not handled. (Reported by Richard Mudgett)
* ASTERISK-25100 - asterisk coredump if host has an IPv6 address that end with ::80 (Reported by Mark Petersen)
* ASTERISK-25122 - Large SIP packet received via pjsip over websocket crashes Asterisk (Reported by Ivan Poddubny)
* ASTERISK-25121 - Stasis: Fix unsafe use of stasis_unsubscribe in modules. (Reported by Corey Farrell)
* ASTERISK-24988 - func_talkdetect: Test is bouncing sporadically (Reported by Joshua Colp)
* ASTERISK-25105 - res_pjsip: Possible incompatibility between qualify_timeout and pjproject-2.4 (Reported by George Joseph)
* ASTERISK-25117 - res_mwi_external_ami: Fix manager action registrations. (Reported by Corey Farrell)
* ASTERISK-25112 - Logger: Configuration settings are not reset to default during reload. (Reported by Corey Farrell)
* ASTERISK-24944 - main/audiohook.c change prevents G722 call recording (Reported by Ronald Raikes)
* ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2 or more digits (Reported by Makoto Dei)
* ASTERISK-25086 - [patch]PJSIP crashes if endpoint missing in Dial() (Reported by snuffy)
* ASTERISK-25089 - res_pjsip_config_wizard: Variable specified in templates aren't being processed correctly (Reported by George Joseph)
* ASTERISK-25090 - CLI core show channel truncates cdr variables (Reported by snuffy)
* ASTERISK-25085 - [patch]Potential crash after unload of func_periodic_hook or test_message (Reported by Corey Farrell)
* ASTERISK-25083 - Message.c: Message channel becomes saturated with frames leading to spammy log messages (Reported by Jonathan Rose)
* ASTERISK-25082 - Asterisk deletes message after doing a playback of an INBOX message using ast_vm_play when the Old folder is full for that mailbox. (Reported by Jonathan Rose)
* ASTERISK-18252 - queue_log mysql time column data format (Reported by Gareth Blades)
* ASTERISK-25041 - [patch]Broken column type checking in res_config_mysql addon (Reported by Alexandre Fournier)
* ASTERISK-21893 - Segfault after call hangup, in ast_channel_hangupcause_set, at channel_internal_api.c (Reported by Aleksandr Gordeev)
* ASTERISK-25074 - Regression: Recent clang-related change broke cross compiling of Asterisk (Reported by Sebastian Kemper)
* ASTERISK-25042 - asterisk.conf options override command-line options. (Reported by Corey Farrell)
* ASTERISK-24442 - Outgoing call files don't work properly when set in the future (Reported by tootai)
* ASTERISK-25057 - res_pjsip_pubsub: Crash in send_notify due to invalid root pointer in sub_tree (Reported by Matt Jordan)
* ASTERISK-24938 - ARI Snoop Channel results in excessive escalating CPU usage (Reported by George Ladoff)
* ASTERISK-25034 - chan_dahdi: Some telco switches occasionally ignore ISDN RESTART requests. (Reported by Richard Mudgett)
* ASTERISK-25003 - Asterisk crashes on attended transfer (using feature) (Reported by Artem Volodin)
* ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always contain waiting time (Reported by Etienne Lessard)
* ASTERISK-25027 - Build System: Many ARI modules are missing dependencies. (Reported by Corey Farrell)
* ASTERISK-25061 - pbx_config: Register manager actions with module version of macro. (Reported by Corey Farrell)
* ASTERISK-25025 - Periodic crashes (in ast_channel_snapshot_create at stasis_channels.c) with Certified Asterisk 13. (Reported by Chet Stevens)
* ASTERISK-25053 - Unit test category /main/presence missing trailing slash. (Reported by Corey Farrell)
* ASTERISK-22708 - res_odbc.conf negative_connection_cache option not respected, failover between DSNs doesn't work (Reported by JoshE)
* ASTERISK-25054 - Formats interface's cannot be unregistered, needs to hold modules until shutdown. (Reported by Corey Farrell)
* ASTERISK-24896 - [patch] Using force black background leads to colours not being reset (Reported by dant)
* ASTERISK-25033 - Asterisk 13 (branch head) won't compile without PJSip (Reported by Peter Whisker)
* ASTERISK-25028 - Build System: Unneeded defines in asterisk/buildopts.h (Reported by Corey Farrell)
* ASTERISK-25048 - Astobj2: Initialization order wrong when both refdebug and AO2_DEBUG are both enabled. (Reported by Corey Farrell)
* ASTERISK-19608 - Asterisk-1.8.x starts rejecting calls with cause code 44 after some time. (Reported by Denis Alberto Martinez)
* ASTERISK-24976 - cdr_odbc not include new columns added on 1.8 (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25037 - res_pjsip_outbound_registration: Potential crash in off-nominal failure case when sending message (Reported by Joshua Colp)
* ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls (Reported by Steve Davies)
* ASTERISK-22790 - check_modem_rate() may return incorrect rate for V.27 (Reported by not here)
* ASTERISK-23231 - Since 405693 If we have res_fax.conf file set to minrate=2400, then res_fax refuse to load (Reported by David Brillert)
* ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400, which is disallowed in res_fax's check_modem_rate (Reported by Matt Jordan)
* ASTERISK-24996 - chan_pjsip: Creating Channel Causes Asterisk to Crash When Duplicate AOR Sections Exist in pjsip.conf (Reported by Ashley Sanders)
* ASTERISK-25020 - Mismatched response to outgoing REGISTER request (Reported by Mark Michelson)
* ASTERISK-25018 - pjsip show endpoints crashes asterisk when qualified aors present (Reported by Ivan Poddubny)
* ASTERISK-24749 - ConfBridge: Wrong language on playing conf-hasjoin and conf-hasleft when played to bridge (Reported by Philippe Bolduc)
* ASTERISK-24845 - pjsip send notify not working with Cisco phone (Reported by Carl Fortin)
* ASTERISK-25004 - Crash in authenticated reinvite after originated T.38 FAX (Reported by Mark Michelson)
* ASTERISK-24999 - PJSIP crashes with malformed contact line (Reported by snuffy)
* ASTERISK-24998 - res_corosync: res_corosync tries to load even if res_corosync.conf is missing (Reported by George Joseph)
* ASTERISK-24997 - Astobj2: Some callers of __adjust_lock do not pre-check the object (Reported by Corey Farrell)
* ASTERISK-24982 - res_pjsip_mwi: Unsolicited MWI NOTIFY only sent on mailbox changes (Reported by Joshua Colp)
* ASTERISK-24991 - Check for ao2_alloc failure in __ast_channel_internal_alloc (Reported by Corey Farrell)
* ASTERISK-24895 - After hangup on the side of the ISDN network no HangupRequest event comes for the dahdi channel. (Reported by Andrew Zherdin)
* ASTERISK-24977 - Contacts that don't use qualify are being marked as unavailable (Reported by George Joseph)
* ASTERISK-24774 - Segfault in ast_context_destroy with extensions.ael and extensions.conf (Reported by Corey Farrell)
* ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have multiple native formats (Reported by Matt Jordan)
* ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build to Fail (Reported by Ashley Sanders)
* ASTERISK-24863 - res_pjsip: No endpoint events raised via AMI when contacts cannot be reached/qualified (Reported by Dmitriy Serov)
* ASTERISK-24869 - Asterisk segfaults on DAHDI attended transfer due to application (appl) being NULL on unbridged channel (Reported by viniciusfontes)
* ASTERISK-24970 - Crash in res_pjsip_pubsub handling of failed notify (Reported by Scott Griepentrog)
* ASTERISK-13721 - memory leak in "strings.c" (Reported by pj)
* ASTERISK-24959 - [patch]CLI command cdr show pgsql status (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-24954 - Git migration: Asterisk version numbers are incompatible with the Test Suite (Reported by Matt Jordan)
* ASTERISK-17608 - func_aes.so cannot be loaded if res_crypto / openssl not compiled (Reported by Warren Selby)
* ASTERISK-24928 - [patch]t38_udptl_maxdatagram in pjsip.conf not honored (Reported by Juergen Spies)
* ASTERISK-24835 - Early Media Not working with Chan SIP and Asterisk 13 (Reported by Andrew Nagy)
* ASTERISK-21777 - Asterisk tries to transcode video instead of audio (Reported by Nick Ruggles)
* ASTERISK-24380 - core: Native formats are set to h264 with certain audio/video codec configuration, resulting in path translation WARNINGs (Reported by Matt Jordan)
* ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken into account (Reported by Frederic Van Espen)
* ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too short (Reported by Y Ateya)
* ASTERISK-24935 - res_pjsip_phoneprov_provider: Fix leaked OBJ_MULTIPLE iterator. (Reported by Corey Farrell)
* ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c (Reported by Vadim)
* ASTERISK-24933 - T38 fails negotiation (Reported by Jonathan Rose)
* ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL byte prefix bug (Reported by Matt Jordan)
* ASTERISK-21211 - chan_iax2 - unprotected access of iaxs[peer->callno] potentially results in segfault (Reported by Jaco Kroon)
* ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working (Reported by Christoph Timm)
* ASTERISK-24910 - "timer=no" and "timer=required" settings in pjsip.conf fail (Reported by Ray Crumrine)
* ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0 (Reported by Jeffrey C. Ollie)
* ASTERISK-24914 - Division by zero in file.c when playback of voicemail with video as h264 (Reported by Marcello Ceschia)
* ASTERISK-24899 - Parking fall-through behavior different in 13 (Reported by Malcolm Davenport)
* ASTERISK-24937 - [patch]res_pjsip_messaging: Messages may be sent out of order (Reported by Mark Michelson)
* ASTERISK-24920 - Asterisk handles duplicate SIP requests as if they were each a new request (Reported by Mark Michelson)
* ASTERISK-24857 - [patch] "timing test", pjsip incoming/outgoing calls, voicemail prompts and recordings all fail when using the kqueue timer source on FreeBSD 10.x (Reported by Justin T. Gibbs)
* ASTERISK-24155 - [patch]Non-portable and non-reliable recursion detection in ast_malloc (Reported by Timo Teräs)
* ASTERISK-24142 - CCSS: crash during shutdown due to device lookup in destroyed container (Reported by David Brillert)
* ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during core restart now (Reported by Peter Katzmann)
* ASTERISK-24805 - [patch] - ASAN: Race condition (heap-use-after-free) on asterisk closing (Reported by Badalian Vyacheslav)
* ASTERISK-24881 - ast_register_atexit should only be used when absolutely needed (Reported by Corey Farrell)
* ASTERISK-24731 - res_pjsip_session cannot be unloaded (Reported by Corey Farrell)
* ASTERISK-24864 - app_confbridge: file playback blocks dtmf (Reported by Kevin Harwell)
* ASTERISK-14233 - [patch] Buddies are always auto-registered when processing the roster (Reported by Simon Arlott)
* ASTERISK-24780 - [patch] - Buddies are always auto-registered when processing the roster (Reported by Simon Arlott)
* ASTERISK-24781 - PJSIP: Unnecessary 180 Ringing messages sent with undesireabe consequences. (Reported by Richard Mudgett)
* ASTERISK-24879 - [patch]Compilation fails due to 64bit time under OpenBSD (Reported by snuffy)
* ASTERISK-24880 - [patch]Compilation under OpenBSD (Reported by snuffy)
* ASTERISK-21765 - [patch] - FILE function's length argument counts from beginning of file rather than the offset (Reported by John Zhong)
* ASTERISK-24817 - init_logger_chain: unreachable code block (Reported by Corey Farrell)
* ASTERISK-24882 - chan_sip: Improve usage of REF_DEBUG (Reported by Corey Farrell)
* ASTERISK-24876 - Investigate reference leaks from tests/channels/local/local_optimize_away (Reported by Corey Farrell)
* ASTERISK-24840 - res_pjsip: conflicting endpoint identifiers (Reported by Kevin Harwell)
* ASTERISK-16779 - Cannot disallow unknown format '' (Reported by Atis Lezdins)
* ASTERISK-18708 - func_curl hangs channel under load (Reported by Dave Cabot)
* ASTERISK-21038 - Bad command completion of "core set debug channel" (Reported by Richard Kenner)
* ASTERISK-19470 - Documentation on app_amd is incorrect (Reported by Frank DiGennaro)
* ASTERISK-24872 - [patch] AMI PJSIPShowEndpoint closes AMI connection on error (Reported by Dmitriy Serov)
* ASTERISK-23666 - CLONE - nested functions aren't portable (Reported by Diederik de Groot)
* ASTERISK-20399 - Compilation on some systems requires the -fnested-functions flag (Reported by David M. Lee)
* ASTERISK-20850 - [patch]Nested functions aren't portable. Adapting RAII_VAR to use clang/llvm blocks to get the same/similar functionality. (Reported by Diederik de Groot)
* ASTERISK-24807 - Missing mandatory field Max-Forwards (Reported by Anatoli)
* ASTERISK-24808 - res_config_odbc: Improper escaping of backslashes occurs with MySQL (Reported by Javier Acosta)
* ASTERISK-23390 - NewExten Event with application AGI shows up before and after AGI runs (Reported by Benjamin Keith Ford)
* ASTERISK-24786 - [patch] - Asterisk terminates when playing a voicemail stored in LDAP (Reported by Graham Barnett)
* ASTERISK-24739 - [patch] - Out of files -- call fails -- numerous files with inodes from under /usr/share/zoneinfo, mostly posixrules (Reported by Ed Hynan)
* ASTERISK-24755 - Asterisk sends unexpected early BYE to transferrer during attended transfer when using a Stasis bridge (Reported by John Bigelow)
* ASTERISK-24830 - res_rtp_asterisk.c checks USE_PJPROJECT not HAVE_PJPROJECT (Reported by Stefan Engström)
* ASTERISK-24825 - Caller ID not recognized using Centrex/Distinctive dialing (Reported by Richard Mudgett)
* ASTERISK-17588 - Caller ID on TDM410P *UK* PSTN (Reported by Daniel Flounders)
* ASTERISK-24838 - chan_sip: Locking inversion occurs when building a peer causes a peer poke during request handling (Reported by Richard Mudgett)
* ASTERISK-24751 - Integer values in json payload to ARI cause asterisk to crash (Reported by jeffrey putnam)
* ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
* ASTERISK-18105 - most of asterisk modules are unbuildable in cygwin environment (Reported by feyfre)
* ASTERISK-21845 - maxcalls exceeded, Asterisk sends out 480 and also BYE (Reported by Tony Ching)
* ASTERISK-15434 - [patch] When ast_pbx_start failed, both an error response and BYE are sent to the caller (Reported by Makoto Dei)
* ASTERISK-23214 - chan_sip WARNING message 'We are requesting SRTP for audio, but they responded without it' is ambiguous and wrong in some cases (Reported by Rusty Newton)
* ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime fail (Reported by Terry Wilson)
* ASTERISK-20233 - SRTP not working with some devices (Eg Grandstream gxv3175) - Message "Can't provide secure audio requested in SDP offer" (Reported by tootai)
* ASTERISK-22748 - SRTP Crypto Offer With Lifetime Not Accepted (Reported by Alejandro Mejia)
* ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid thread ID being passed to pthread_kill (Reported by JoshE)
* ASTERISK-24812 - ARI: Creating channels through /channels resource always uses SLIN, which results in unneeded transcoding (Reported by Matt Jordan)
* ASTERISK-24797 - bridge_softmix: G.729 codec license held (Reported by Kevin Harwell)
* ASTERISK-24677 - ARI GET variable on channel provides unhelpful response on non-existent variable (Reported by Joshua Colp)
* ASTERISK-24785 - 'Expires' header missing from 200 OK on REGISTER (Reported by Ross Beer)
* ASTERISK-24499 - Need more explicit debug when PJSIP dialstring is invalid (Reported by Rusty Newton)
* ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML (Reported by Ashley Sanders)
* ASTERISK-24796 - Codecs and bucket schema's prevent module unload (Reported by Corey Farrell)
* ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc OSX with 64 bit integers (Reported by Corey Farrell)
* ASTERISK-24787 - [patch] - Microsoft exchange incompatibility for playing back messages stored in IMAP - play_message: No origtime (Reported by Graham Barnett)
* ASTERISK-22670 - Asterisk crashes when processing ISDN AoC Events (Reported by klaus3000)
* ASTERISK-24689 - Segfault on hangup after outgoing PRI-Euroisdn call (Reported by Marcel Manz)
* ASTERISK-24740 - [patch]Segmentation fault on aoc-e event (Reported by Panos Gkikakis)
* ASTERISK-24799 - [patch] make fails with undefined reference to SSLv3_client_method (Reported by Alexander Traud)
* ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove (Reported by Corey Farrell)
* ASTERISK-24700 - CRASH: NULL channel is being passed to ast_bridge_transfer_attended() (Reported by Zane Conkle)
* ASTERISK-24791 - Crash in ast_rtcp_write_report (Reported by JoshE)
* ASTERISK-24085 - Documentation - We should remove or further document the 'contact' section in pjsip.conf (Reported by Rusty Newton)
* ASTERISK-24632 - install_prereq script installs pjproject without IPv6 support (Reported by Rusty Newton)
* ASTERISK-24685 - "pjsip show version" CLI command (Reported by Joshua Colp)
* ASTERISK-24768 - res_timing_pthread: file descriptor leak (Reported by Matthias Urlichs)
* ASTERISK-24612 - res_pjsip: No information if a required sorcery wizard is not loaded (Reported by Joshua Colp)
* ASTERISK-24716 - Improve pjsip log messages for presence subscription failure (Reported by Rusty Newton)
* ASTERISK-24771 - ${CHANNEL(pjsip)} - segfault (Reported by Niklas Larsson)
* ASTERISK-24727 - PJSIP: Crash experienced during multi-Asterisk transfer scenario. (Reported by Mark Michelson)
* ASTERISK-24015 - app_transfer fails with PJSIP channels (Reported by Private Name)
* ASTERISK-24741 - dtls_handler causes Asterisk to crash (Reported by Zane Conkle)
* ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to fully disconnect underlying socket, leading to events being dropped with no additional information (Reported by Matt Jordan)
* ASTERISK-24752 - Crash in bridge_manager_service_req when bridge is destroyed by ARI during shutdown (Reported by Richard Mudgett)
* ASTERISK-24772 - ODBC error in realtime sippeers when device unregisters under MariaDB (Reported by Richard Miller)
* ASTERISK-24479 - Enable REF_DEBUG for module references (Reported by Corey Farrell)
* ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in res_odbc (Reported by ibercom)
* ASTERISK-24769 - res_pjsip_sdp_rtp: Local ICE candidates leaked (Reported by Matt Jordan)
* ASTERISK-24748 - res_pjsip: If wizards explicitly configured in sorcery.conf false ERROR messages may occur (Reported by Joshua Colp)
* ASTERISK-24616 - Crash in res_format_attr_h264 due to invalid string copy (Reported by Yura Kocyuba)
* ASTERISK-24737 - When agent not logged in, agent status shows unavailable, queue status shows agent invalid (Reported by Richard Mudgett)
* ASTERISK-24635 - PJSIP outbound PUBLISH crashes when no response is ever received (Reported by Marco Paland)
* ASTERISK-24736 - Memory Leak Fixes (Reported by Mark Michelson)
* ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by Stephan Eisvogel)
* ASTERISK-24711 - DTLS handshake broken with latest OpenSSL versions (Reported by Jared Biel)
* ASTERISK-24666 - Security Vulnerability: RTP not closed after sip call using unsupported codec (Reported by Y Ateya)
* ASTERISK-24676 - Security Vulnerability: URL request injection in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
* ASTERISK-24729 - Outbound registration not occuring on new registrations after reload. (Reported by Richard Mudgett)
* ASTERISK-24728 - tcptls: Bad file descriptor error when reloading chan_sip (Reported by Kevin Harwell)
* ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported by Kevin Harwell)
* ASTERISK-24485 - res_pjsip cannot be unloaded or shutdown (Reported by Corey Farrell)
* ASTERISK-24719 - ConfBridge recording channels get stuck when recording started/stopped more than once (Reported by Richard Mudgett)
* ASTERISK-24721 - manager: ModuleLoad action incorrectly reports 'module not found' during a Reload operation (Reported by Matt Jordan)
* ASTERISK-24723 - confbridge: CLI command 'confbridge list XXXX' no longer displays user menus (Reported by Matt Jordan)
* ASTERISK-24539 - Compile fails on OSX because of sem_timedwait in bridge_channel.c (Reported by George Joseph)
* ASTERISK-24544 - Compile fails on OSX Yosemite because of incorrect detection of htonll and ntohll (Reported by George Joseph)
* ASTERISK-24231 - crash: CLI execution of realtime destroy sippeers id 1 causes crash due to NULL name provided to ast_variable (Reported by Niklas Larsson)
* ASTERISK-24626 - Voicemail passwords not being stored in ARA (Reported by Paddy Grice)
* ASTERISK-24693 - Investigate and fix memory leaks in Asterisk (Reported by Kevin Harwell)
* ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive column comparison for 'defaultuser' (Reported by HZMI8gkCvPpom0tM)
* ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor m() option does not queue an MWI event (Reported by Gareth Palmer)
* ASTERISK-24673 - outgoing sip registers cannot be removed or modified without doing restart (or doing module unload chan_sip.so) (Reported by Stefan Engström)
* ASTERISK-24640 - Registration pending stays forever after sip reload (Reported by Max Man)
* ASTERISK-24682 - app_dial: Multiple DialEnd events emitted when MACRO_RESULT or GOSUB_RESULT are an unexpected value (Reported by Matt Jordan)
* ASTERISK-24560 - Creating a named ARI bridge twice causes a crash (Reported by Kinsey Moore)
* ASTERISK-24600 - Stuck IAX channels, Asterisk stops responding to most traffic, potential deadlock (Reported by Jeff Collell)
* ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects 32-bit packages on 64-bit hosts (Reported by Ben Klang)
* ASTERISK-24288 - [patch] - ODBC usage with app_voicemail - voicemail is not deleted after review, hangup (Reported by LEI FU)
* ASTERISK-24615 - When Multiple Transports Exist in pjsip.conf, Incorrect External Addresses is Used in SIP Packets When Responding to INVITE (Reported by David Justl)
* ASTERISK-24624 - Transfer to invalid extension results in hung channel. (Reported by Zane Conkle)
* ASTERISK-24663 - [patch] Unnamed semaphore autoconf check fails on cross compilation (Reported by abelbeck)
* ASTERISK-24655 - res_pjsip_outbound_publish: Hang on shutdown while attempting to publish (Reported by Kevin Harwell)
* ASTERISK-23991 - [patch]asterisk.pc file contains a small error in the CFlags returned (Reported by Diederik de Groot)
* ASTERISK-23850 - Park Application does not respect Return Context Priority (Reported by Andrew Nagy)
* ASTERISK-24665 - Configure check required for pjsip_get_dest_info() (Reported by Mark Michelson)
* ASTERISK-24049 - Asterisk Manager Interface: A number of list type responses aren't using astman_send_listack (Reported by Jonathan Rose)
* ASTERISK-20744 - [patch] Security event logging does not work over syslog (Reported by Michael Keuter)
* ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT (Reported by Kristian Høgh)
* ASTERISK-24474 - sip_to_pjsip.py lacks documentation and does not function (Reported by John Kiniston)
* ASTERISK-24637 - Channel re-enters Stasis() when it should not (Reported by John Bigelow)
* ASTERISK-24591 - Stasis() side of an ARI originated channel cannot be Redirected (Reported by Kinsey Moore)
* ASTERISK-24376 - res_pjsip_refer: REFER request for remote session attempts to direct channel to external_replaces extension instead of context, without providing for the Referred-To SIP URI (Reported by Matt Jordan)
* ASTERISK-24513 - Local channel apparently leaked in off-nominal DTMF attended transfer (Reported by Mark Michelson)
* ASTERISK-24267 - Queue variables associated with setinterfacevar, setqueueentryvar, setqueuevar are not passed to local channel (Reported by Mitch Claborn)
* ASTERISK-23841 - DTMF atxfer doesn't set CallerID for the recall calls to the transferrer. (Reported by Richard Mudgett)
* ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong destination when 'sendrpid=yes' (in proxy environment) (Reported by Karsten Wemheuer)
* ASTERISK-23733 - 'reload acl' fails if acl.conf is not present on startup (Reported by Richard Kenner)
* ASTERISK-24566 - Uninit buf in WS write (Reported by Badalian Vyacheslav)
* ASTERISK-24337 - Spammy DEBUG message needs to be at a higher level - 'Remote address is null, most likely RTP has been stopped' (Reported by Rusty Newton)
* ASTERISK-24459 - bridge_native_rtp: Native RTP bridging is chosen for RTP compatible channels when the DTMF mode is not compatible (Reported by Yaniv Simhi)
* ASTERISK-24536 - AMI redirect with PJSIP fails to move extra channel (Reported by Niklas Larsson)
* ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly casts char to unsigned int (Reported by Walter Doekes)
* ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is enabled (Reported by Andreas Steinmetz)
* ASTERISK-22455 - Asterisk 12 on Ubuntu Lucid deadlocks with DEBUG_THREADS+OPTIONAL_API enabled (Reported by David M. Lee)
* ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag enabled (Reported by Richard Mudgett)
* ASTERISK-24604 - res_rtp_asterisk: Crash during restart due to race condition in accessing codec in stored ast_frame and codec core (Reported by Matt Jordan)
* ASTERISK-24563 - Direct Media calls within private network sometimes get one way audio (Reported by Kevin Harwell)
* ASTERISK-24607 - res_pjsip_session: re-INVITE with declined media streams results in 488 (Reported by Matt Jordan)
* ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS from JSSIP (Reported by Badalian Vyacheslav)
* ASTERISK-24514 - res_pjsip_outbound_registration: stack overflow when using non-default sorcery wizard (Reported by Kevin Harwell)
* ASTERISK-24342 - PJSIP: Qualifying endpoints attempts to do them all at the same time. (Reported by Richard Mudgett)
Improvements made in this release:
-----------------------------------
* ASTERISK-26088 - Investigate heavy memory utilization by res_pjsip_pubsub (Reported by Richard Mudgett)
* ASTERISK-25930 - PJSIP: disable multi domain to improve realtime performace (Reported by Alexei Gradinari)
* ASTERISK-25495 - [patch] Prevent old-update packages on repository Debian systems (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25846 - Gracefully deal with Absent Stasis Apps (Reported by Andrew Nagy)
* ASTERISK-25791 - res_pjsip_caller_id: Lack of support for Anonymous
* ASTERISK-24813 - asterisk.c: #if statement in listener() confuses code folding editors (Reported by Corey Farrell)
* ASTERISK-25767 - [patch] Add check to configure for sanitizes (Reported by Badalian Vyacheslav)
* ASTERISK-25068 - Move commonly used FreePBX extra sounds to the core set (Reported by Rusty Newton)
* ASTERISK-25627 - Easily Preventable Compile Warning (Reported by Diederik de Groot)
* ASTERISK-25618 - res_pjsip: Check for readability of TLS files at startup (Reported by George Joseph)
* ASTERISK-25572 - Endpoints: Add StatsD stats for Asterisk endpoints (Reported by Matt Jordan)
* ASTERISK-25571 - PJSIP: Add StatsD stats for some common PJSIP objects (Reported by Matt Jordan)
* ASTERISK-25518 - taskprocessor: Add high water mark (Reported by Jonathan Rose)
* ASTERISK-25477 - pjsip show "command" like [criteria] (Reported by Bryant Zimmerman)
* ASTERISK-24718 - [patch]Add inital support of "sanitize" to configure (Reported by Badalian Vyacheslav)
* ASTERISK-24870 - ARI: Subscriptions to bridges generally not super useful (Reported by Matt Jordan)
* ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init() defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
* ASTERISK-25256 - [patch]Post AMI VarSet to empty string events when Asterisk deletes a dialplan variable. (Reported by Richard Mudgett)
* ASTERISK-25067 - Sorcery Caching: Implement a new caching module (Reported by Matt Jordan)
* ASTERISK-25040 - pbx: Improve performance of reloads by making hint destruction more performant (Reported by Matt Jordan)
* ASTERISK-25114 - res_pjsip: Add AMI events for chan_pjsip contact lifecycle changes (Reported by George Joseph)
* ASTERISK-25072 - res_pjsip_outbound_registration: line functionality. Additional check for using the request URI (Reported by Dmitriy Serov)
* ASTERISK-25044 - sorcery: Add ability to insert a new wizard into an object type's list (Reported by George Joseph)
* ASTERISK-24892 - Super Awesome Company sound prompts (Reported by Rusty Newton)
* ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove Hjelm)
* ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL (Reported by Alexander Traud)
* ASTERISK-25045 - vector: Add new capabilities and unit tests (Reported by George Joseph)
* ASTERISK-24706 - [patch]add auto-dtmf mode for pjsip (Reported by yaron nahum)
* ASTERISK-25051 - Remove unneeded uses of optional_api providers. (Reported by Corey Farrell)
* ASTERISK-24917 - [patch] clang compilation warnings (Reported by Diederik de Groot)
* ASTERISK-24949 - res_pjsip_outbound_registration: Backport line functionality (Reported by Joshua Colp)
* ASTERISK-24965 - cel_pgsql - log_error string references CDR instead of CEL (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-24918 - pjsip: add CLI options to display global and system configuration (Reported by Scott Griepentrog)
* ASTERISK-24862 - [patch] Support in-dialog OPTIONS (Reported by yaron nahum)
* ASTERISK-24802 - stasis: set a channel variable on websocket disconnect error (Reported by Kevin Harwell)
* ASTERISK-24133 - [patch]Please support Clang; Allow no-exec stacks (Reported by Jeffrey Walton)
* ASTERISK-24790 - Reduce spurious noise in logs from voicemail - Couldn't find mailbox %s in context (Reported by Graham Barnett)
* ASTERISK-24811 - asterisk-publication sorcery object does not use realtime (Reported by Matt Hoskins)
* ASTERISK-24745 - [patch]Add no_answer to ARI hangup causes (Reported by Ben Merrills)
* ASTERISK-24316 - For httpd server, need option to define server name for security purposes (Reported by Andrew Nagy)
* ASTERISK-24671 - Missing docs for the CDR AMI Event (Reported by Dan Jenkins)
* ASTERISK-24575 - [patch]Make capath work for res_pjsip (Reported by cloos)
* ASTERISK-24678 - [PATCH] Added atxfer * settings to features.conf.sample (Reported by Niklas Larsson)
* ASTERISK-24412 - [patch]Incomplete channel originate/continue handling with ARI (Reported by Nir Simionovich (GreenfieldTech - Israel))
* ASTERISK-24643 - res_pjsip: Add user=phone option (Reported by Matt Jordan)
* ASTERISK-24644 - res_pjsip_keepalive: Add keepalive module for connection-oriented transports. (Reported by Matt Jordan)
* ASTERISK-24553 - ARI/AMI: Include language in standard channel snapshot output (Reported by Matt Jordan)
* ASTERISK-24552 - ARI: Allow associating a channel as an initiator of an Origination for record keeping purposes (Reported by Matt Jordan)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.8-cert1