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