The Asterisk Development Team has announced the first beta of Asterisk 14.0.0. This beta is available for immediate

download at

The release of Asterisk 14.0.0-beta1 resolves several issues reported by the community and would have not been possible without your participation.

Thank you!

The following are the issues resolved in this beta:

New Features made in this release:


* ASTERISK-25904 – PJSIP: add contact.updated event (Reported by

     Alexei Gradinari)

* ASTERISK-26058 – [Patch] Add uptime and last reloaded to

     FullyBooted AMI event (Reported by Niklas Larsson)

* ASTERISK-25925 – Allow Early Bridges on ARI Dials (Reported by

     Mark Michelson)

* ASTERISK-26068 – Multicast RTP Options (Reported by Mark


* ASTERISK-26042 – ARI: Allow downloading of the media associated

     with a stored recording (Reported by Matt Jordan)

* ASTERISK-25425 – logger: Add JSON structured logging (Reported

     by Matt Jordan)

* ASTERISK-25900 – PJSIP Endpoint IP Access Controls (Reported by

     Alexei Gradinari)

* ASTERISK-25972 – res_pjsip_exten_state: Use body generator to

     publish extension state (Reported by Richard Mudgett)

* ASTERISK-25889 – ARI: Add separate “create” and “dial”

     operations for channels (Reported by Mark Michelson)

* ASTERISK-25803 – chan_sip: Optionally supply

     fromuser/fromdomain in SIP dial string (Reported by Walter


* 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


* ASTERISK-25660 – Add sipp-sendfax.xml and to

     contrib/scripts. (Reported by Walter Doekes)

* ASTERISK-25591 – Complete List of Header Files

     (#include): iwyu (Reported by Alexander Traud)

* ASTERISK-25551 – Ability to add channel to an existing

     bridge by specifying an existing channel prefix (Reported by

     Alec Davis)

* ASTERISK-25419 – Dialplan Application for Integration of StatsD

     (Reported by Ashley Sanders)

* ASTERISK-25549 – Confbridge: Add participant timeout option

     (Reported by Mark Michelson)

* ASTERISK-24922 – ARI: Add the ability to intercept hold and

     raise an event (Reported by Matt Jordan)

* ASTERISK-25479 – Allow CDR’s to be modified before being

     dispatched to engines (Reported by Jonh Wendell)

* ASTERISK-25480 – Add field PauseReason on

     QueueMemberStatus (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-25377 – res_pjsip: Change default “From user” from UUID

     to something more palatable (Reported by Mark Michelson)

* ASTERISK-25252 – ARI: Add the ability to manipulate log channels

     (Reported by Matt Jordan)

* ASTERISK-25259 – chan_pjsip: Add rtptimeout support (Reported by

     Joshua Colp)

* ASTERISK-25238 – ARI: Support push configuration (Reported by

     Matt Jordan)

* ASTERISK-25173 – ARI: Add the ability to load/reload/unload an

     Asterisk module (Reported by Matt Jordan)

* ASTERISK-25006 – Add support set character for quoted

     identifiers  (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-23186 – Add usegmtime option to cel_pgsql

     (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-24931 – dns: Add support for SRV records. (Reported by

     Joshua Colp)

* ASTERISK-24834 – DNS Overhaul: Implement the proposed core API –

     sync/async functions, resolver registration (Reported by Matt


* ASTERISK-24836 – DNS Overhaul: Write a Resolver Implementation

     (Reported by Matt Jordan)

* ASTERISK-22591 – Prevent Asterisk from writing received

     SMS content in log (Reported by Jan Juergens)

* ASTERISK-17899 – Handle crypto lifetime in SDES-SRTP negotiation

     (Reported by Dwayne Hubbard)

* ASTERISK-24703 – ARI: Add the ability to “transfer” (redirect) a

     channel (Reported by Matt Jordan)

* ASTERISK-24363 – Add ability for Channel Drivers to

     provide Presence State information (Reported by Gareth Palmer)

* ASTERISK-24554 – AMI/ARI: Generate events on connected line

     changes (Reported by Matt Jordan)

* ASTERISK-24276 – [Patch] Option to make app MOH override channel

     musicclass (Reported by Kristian Høgh)

* ASTERISK-23871 – RLS Tests: Implement RLS off-nominal tests

     (Reported by Mark Michelson)

* ASTERISK-23823 – Option to keep queuerules in realtime

     (Reported by Michael K.)

Bugs fixed in this release:


* ASTERISK-26227 – sqlalchemy error due to long identifier name

     (Reported by Mark Michelson)

* ASTERISK-26221 – chan_sip: iLBC does not include correct mode

     (Reported by Aaron Meriwether)

* ASTERISK-26216 – res_fax: Deadlock when detect fax while channel

     executing Playback (Reported by Richard Mudgett)

* ASTERISK-26214 – Allow arbitrary time for fax detection to end

     on a channel (Reported by Richard Mudgett)

* ASTERISK-23013 – Deadlock between ‘sip show channels’

     command and attended transfer handling (Reported by Ben


* ASTERISK-26212 – Makefile: Retain XML Declaration and

     DTD in docs. (Reported by Alexander Traud)

* ASTERISK-26211 – Unit tests: AST_TEST_DEFINE should be used in

     conditional code. (Reported by Corey Farrell)

* ASTERISK-26207 – sRTP: Count a roll-over of the sequence

     number even on lost packets. (Reported by Alexander Traud)

* ASTERISK-26038 – ‘make install’ doesn’t seem to install OS/X

     init files (Reported by Tzafrir Cohen)

* ASTERISK-26133 – app_queue: Queue members receive multiple calls

     (Reported by Richard Miller)

* ASTERISK-26196 – pbx: Time based includes can leak timezone

     string (Reported by Corey Farrell)

* ASTERISK-26193 – chan_sip: reference leak in mwi_event_cb

     (Reported by Corey Farrell)

* ASTERISK-26191 – threadpool: Leak on duplicate taskprocessor for

     ast_threadpool_serializer_group (Reported by Corey Farrell)

* ASTERISK-25659 – res_rtp_asterisk: ECDH not negotiated causing

     DTLS failure occurred on RTP instance (Reported by Edwin


* ASTERISK-26046 – Avoid obsolete warnings on autoconf.

     (Reported by Alexander Traud)

* ASTERISK-26160 – pjsip: Updated->Reachable during qualify

     (Reported by Matt Jordan)

* ASTERISK-26177 – func_odbc: Database handle is kept when it

     should be released (Reported by Leandro Dardini)

* ASTERISK-25289 – Build System does not respect CFLAGS and

     CXXFLAGS when building menuselect (Reported by Jeffrey Walton)

* ASTERISK-26119 – fix: memory leaks, resource leaks, out

     of bounds and bugs (Reported by Alexei Gradinari)

* ASTERISK-26184 – chan_sip: Reference leaks in error paths.

     (Reported by Corey Farrell)

* ASTERISK-26181 – REF_DEBUG: Node object incorrectly logged

     during duplicate replacement (Reported by Corey Farrell)

* ASTERISK-26179 – chan_sip: Second T.38 request fails (Reported

     by Joshua Colp)

* ASTERISK-26180 – PJSIP: provide valid tcp nodelay option for

     reuse (Reported by Scott Griepentrog)

* ASTERISK-25772 – res_pjsip: Unexpected two BYE when answered

     (Reported by Dmitriy Serov)

* ASTERISK-26099 – res_pjsip_pubsub: Crash when sending request

     due to server timeout (Reported by Ross Beer)

* ASTERISK-26144 – Crash on loading codecs g729/g723 (Reported by

     Alexei Gradinari)

* ASTERISK-26157 – Build:   Fix errors highlighted by GCC 6.x

     (Reported by George Joseph)

* ASTERISK-26021 – Build codecs siren7 and siren14 for Asterisk 13

     (Reported by Daniel Denson)

* ASTERISK-26141 – res_fax: fax_v21_session_new leaks reference to

     v21_details (Reported by Corey Farrell)

* ASTERISK-26140 – res_rtp_asterisk: gcc 6 caught a

     self-comparison (Reported by George Joseph)

* ASTERISK-26138 – chan_unistim:  Under FreeBSD, chan_unistim

     generates a compile error (Reported by George Joseph)

* ASTERISK-26128 – Alembic scripts are failing (Reported by Mark


* ASTERISK-26139 – test_res_pjsip_scheduler:  Compile failure if

     pjproject isn’t installed in a system location (Reported by

     George Joseph)

* ASTERISK-26130 – WebRTC: Should use latest DTLS version.

     (Reported by Alexander Traud)

* ASTERISK-26132 – PJSIP: provide transport type with received

     messages (Reported by Scott Griepentrog)

* ASTERISK-26127 – res_pjsip_session: Crash due to race condition

     between res_pjsip_session unload and timer (Reported by Joshua


* ASTERISK-26045 – app_voicemail: fix bugs, imap mm_status

     log change to debug (Reported by Alexei Gradinari)

* ASTERISK-26083 – ARI: Announcer channels staying around after

     playback to a bridge is finished (Reported by Per Jensen)

* ASTERISK-26126 – leverage ‘bindaddr’ for TLS in

     http.conf (Reported by Alexander Traud)

* ASTERISK-26097 – CLI: show maximum file descriptors

     (Reported by Alexander Traud)

* ASTERISK-25262 – Memory leak when a caller channel does multiple

     dials and CEL is enabled (Reported by Etienne Lessard)

* ASTERISK-26047 – ARI allows certain commands to run on down

     channels. (Reported by Mark Michelson)

* ASTERISK-25959 –

     http_media_cache/retrieve_cache_control_directives: Sporadic

     failure (Reported by Joshua Colp)

* ASTERISK-26103 – cdr:  Assert on ‘dial end’ event during a blond

     transfer (Reported by George Joseph)

* ASTERISK-26092 – [Segfault] in res_rtp_asterisk.c:4268 after

     Remotely bridged channels (Reported by Niklas Larsson)

* ASTERISK-26089 – Invalid security events during boot using PJSIP

     Realtime (Reported by Scott Griepentrog)

* ASTERISK-26096 – res_hep: Crash when configuration file is

     missing (Reported by Niklas Larsson)

* ASTERISK-26074 – res_odbc: Deadlock within UnixODBC (Reported by

     Ross Beer)

* ASTERISK-26054 – Asterisk crashes (core dump) (Reported by B.


* ASTERISK-26069 – Asterisk truncates To: header, dropping the

     closing ‘>’ (Reported by Vasil Kolev)

* ASTERISK-24436 – Missing header in res/res_srtp.c when compiling

     against libsrtp-1.5.0 (Reported by Patrick Laimbock)

* ASTERISK-26091 – ar cru creates warning, instead use ar

     cr (Reported by Alexander Traud)

* ASTERISK-26070 – ari/channels:  Creating a local channel without

     an originator adds all audio formats to it’s capabilities

     (Reported by George Joseph)

* ASTERISK-26078 – core: Memory leak in logging (Reported by

     Etienne Lessard)

* ASTERISK-26065 – chan_pjsip: MWI NOTIFY contents not ordered

     properly (Reported by Ross Beer)

* ASTERISK-26063 – ${PJSIP_HEADER(read,Call-ID)} does not work –

     documentation needs clarification for when read/write is

     possible (Reported by Private Name)

* ASTERISK-25777 – data race in threadpool (Reported by Badalian


* ASTERISK-26053 – res_pjsip_outbound_publish: Crash when shutting

     down (Reported by Joshua Colp)

* ASTERISK-26049 – res_pjsip: Crash when our own request timer

     fires (Reported by Joshua Colp)

* ASTERISK-25669 – CURL incorrect trim for non ASCII

     characters (Reported by Jesper)

* ASTERISK-26029 – parking: ast_parking_park_call should return

     parking_space instead of parking_exten (Reported by Diederik de


* ASTERISK-25938 – res_odbc: MySQL/MariaDB statement

     LAST_INSERT_ID() always returns zero. (Reported by Edwin


* ASTERISK-25941 – chan_pjsip: Crash on an immediate SIP final

     response (Reported by Javier Riveros )

* ASTERISK-26014 – res_sorcery_astdb: Make tolerant of unknown

     fields (Reported by Joshua Colp)

* ASTERISK-24986 – keepalive INFO packages ignored by asterisk

     (Reported by Ilya Trikoz)

* ASTERISK-26034 – T.38 passthrough problem behind firewall due to

     early nosignal packet (Reported by George Joseph)

* ASTERISK-26030 – call cut because of double Session-Expires

     header in re-invite after proxy authentication is required

     (Reported by George Joseph)

* ASTERISK-25964 – Outbound registrations created via ARI/push

     configuration do not clean up outbound registrations currently

     in flight (Reported by Matt Jordan)

* ASTERISK-26005 – res_pjsip: Multiple SIP messages are combined

     into 1 TCP packet (Reported by Ross Beer)

* ASTERISK-25352 – res_hep_rtcp correlation_id is different then

     res_hep (Reported by Kevin Scott Adams)

* ASTERISK-26007 – res_pjsip: Endpoints deleting early after

     upgrade from 13.8.2 to 13.9 (Reported by Greg Siemon)

* ASTERISK-25990 – PJSIP TLS registration should respect

     client_uri scheme when generating Contact URI (Reported by

     Sebastian Damm)

* ASTERISK-26008 – app_followme does not delete recorded name

     prompt (Reported by Tzafrir Cohen)

* ASTERISK-25978 – res_pjsip_authenticator_digest: Should not use

     source port in nonce verification (Reported by Mark Michelson)

* ASTERISK-26004 – res_pjsip:  The transport/method parameter is

     ignored (Reported by George Joseph)

* ASTERISK-25999 – res_pjsip_dialog_info_body_generator: Remove

     subscription requirement (Reported by Joshua Colp)

* ASTERISK-25993 – pjproject: Allow bundling to not require

     everything it does (Reported by Joshua Colp)

* ASTERISK-25998 – file: Crash when using nativeformats (Reported

     by Joshua Colp)

* ASTERISK-25826 – PJSIP / Sorcery slow load from realtime

     (Reported by Ross Beer)

* ASTERISK-25956 – Compilation error in conditionally compiled

     code in config_options.c (Reported by Chris Trobridge)

* ASTERISK-25968 – pjproject_bundled:  Configure and make need to

     be re-tested (Reported by George Joseph)

* ASTERISK-24463 – Voicemail email address corrupt or not sent

     when message is in the process of being recorded during reload

     (Reported by John Campbell)

* ASTERISK-25922 – res_pjsip_exten_state: Add configuration

     support for publishing (Reported by Joshua Colp)

* ASTERISK-25970 – Segfault in pjsip_url_compare (Reported by

     Dmitriy Serov)

* ASTERISK-25963 – func_odbc requires reconnect checks for stale

     connections (Reported by Ross Beer)

* ASTERISK-25961 – tests/channels/SIP/sip_tls_call: Sporadic crash

     when running test (Reported by Joshua Colp)

* ASTERISK-16115 – problem with ringinuse=no, queue

     members receive sometimes two calls (Reported by nik600)

* ASTERISK-25917 – app_voicemail: passwordlocation=spooldir

     only works if you manually add secret.conf yourself (Reported by

     Jonathan R. Rose)

* ASTERISK-25954 – Manager QueueSummary and QueueStatus Actions

     are case sensitive to QueueName (Reported by Javier Acosta)

* ASTERISK-25951 – res_agi:  run_agi eats frames it shouldn’t

     (Reported by George Joseph)

* ASTERISK-25950 – SIP channel does not send PeerStatus

     events for autocreated peers (Reported by Kirill Katsnelson)

* ASTERISK-25927 – Removed option “registertrying” is still

     documented in sip.conf.sample (Reported by Etienne Lessard)

* ASTERISK-25947 – Protocol transfers to stasis applications are

     missing the StasisStart with the replace_channel object.

     (Reported by Richard Mudgett)

* ASTERISK-24649 – Pushing of channel into bridge fails; Stasis

     fails to get app name (Reported by John Bigelow)

* ASTERISK-24782 – StasisEnd event not present for channel that

     was swapped out for another after completing attended transfer

     (Reported by John Bigelow)

* ASTERISK-25942 – res_pjsip_caller_id: Transfer results in mixed

     ConnectedLine information (Reported by George Joseph)

* ASTERISK-25928 – res_pjsip: URI validation done outside of PJSIP

     thread (Reported by Joshua Colp)

* ASTERISK-25929 – res_pjsip_registrar: AOR_CONTACT_ADDED events

     not raised (Reported by Joshua Colp)

* ASTERISK-25934 – chan_sip should not require sipregs or

     updateable sippeers table unless rt (Reported by Jaco Kroon)

* ASTERISK-25888 – Frequent segfaults in function can_ring_entry()

     of app_queue.c (Reported by Sébastien Couture)

* ASTERISK-25796 – res_pjsip: DOS/Crash when TCP/TLS sockets

     exceed pjproject PJ_IOQUEUE_MAX_HANDLES (Reported by George


* ASTERISK-25707 – Long contact URIs or hostnames can crash

     pjproject/Asterisk under certain conditions (Reported by George


* ASTERISK-25123 – Bracketed IPv6 Contact header parameter

     unparsable with Asterisk/PJSIP (Reported by Anthony Messina)

* ASTERISK-25874 – app_voicemail: Stack buffer overflow in

     test_voicemail_notify_endl (Reported by Badalian Vyacheslav)

* ASTERISK-25912 – chan_local passes AST_CONTROL_PVT_CAUSE_CODE

     without adding them to the local hangupcauses via

     ast_channel_hangupcause_hash_set (Reported by Jaco Kroon)

* ASTERISK-25885 – res_pjsip: Race condition between adding

     contact and automatic expiration (Reported by Joshua Colp)

* ASTERISK-25910 – pjproject:  Via headers are not parsed when

     “received” contains an IPv6 address (Reported by George Joseph)

* ASTERISK-25890 – Asterisk 13.8.0 alembic database update fails

     (Reported by Harley Peters)

* ASTERISK-25894 – webrtc video broken due to missing

     marker bits in RTP streams (Reported by Jacek Konieczny)

* ASTERISK-25881 – pbx: Add support for autohints (Reported by

     Joshua Colp)

* ASTERISK-25854 – No audio after HOLD/RESUME – incorrect

     a=recvonly in SDP from Asterisk (Reported by Robert McGilvray)

* ASTERISK-25868 – Sorcery “append to category” should allow

     filters (Reported by Nick Repin)

* ASTERISK-25873 – res_pjsip: Bundled pjproject: compile error,

     cannot find -lasteriskpj (Reported by Hans van Eijsden)

* ASTERISK-25882 – ARI: Crash can occur due to race condition when

     attempting to operate on a hung up channel (Part 2) (Reported by

     Richard Mudgett)

* ASTERISK-25867 – Video delay on app_echo (Reported by

     Jacek Konieczny)

* ASTERISK-24605 – res_parking option parkeddynamic does not work

     with the core Features ‘parkcall’ (DTMF initiated parking)

     (Reported by Philip Correia)

* ASTERISK-24596 – Unclear how to use Park application with

     res_parking ‘parkeddynamic’ enabled. Documentation? (Reported by

     Philip Correia)

* ASTERISK-25825 – Crashes during shutdown when running CLI

     commands (Reported by Mark Michelson)

* ASTERISK-24543 – Asterisk 13 responds to SIP Invite with all

     possible codecs configured for peer as opposed to intersection

     of configured codecs and offered codecs (Reported by Taylor


* ASTERISK-25407 – Asterisk fails to log to multiple syslog

     destinations (Reported by Elazar Broad)

* ASTERISK-25510 – Log to syslog failing (Reported by

     Michael Newton)

* ASTERISK-25857 – func_aes: incorrect use of strlen() leads to

     data corruption (Reported by Gianluca Merlo)

* ASTERISK-25849 – chan_pjsip: transfers with direct media

     sometimes drops audio (Reported by Kevin Harwell)

* ASTERISK-25814 – Segfault at f ip in

     (Reported by Sergio Medina Toledo)

* ASTERISK-25023 – Deadlock in chan_sip in

     update_provisional_keepalive (Reported by Arnd Schmitter)

* ASTERISK-25321 – 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


* ASTERISK-25582 – Testsuite: Reactor timeout error in

     tests/fax/pjsip/directmedia_reinvite_t38 (Reported by Matt


* ASTERISK-25811 – Unable to delete object from sorcery cache

     (Reported by Ross Beer)

* ASTERISK-25800 – 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


* ASTERISK-25721 – res_phoneprov: memory leak and

     heap-use-after-free (Reported by Badalian Vyacheslav)

* ASTERISK-25272 – 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 Moučka)

* ASTERISK-20987 – non-admin users, who join muted conference are

     not being muted (Reported by hristo)

* ASTERISK-25737 – res_pjsip_outbound_registration: line option

     not in Alembic (Reported by Joshua Colp)

* ASTERISK-24972 – Transport Layer Security (TLS) Protocol BEAST

     Vulnerability – Investigate vulnerability of HTTP server

     (Reported by Alex A. Welzl)

* ASTERISK-25603 – udptl: Uninitialized lengths and bufs in

     udptl_rx_packet cause ast_frdup crash (Reported by Walter


* ASTERISK-25742 – Secondary IFP Packets can result in accessing

     uninitialized pointers and a crash (Reported by Torrey Searle)

* ASTERISK-25397 – 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


* ASTERISK-25714 – ASAN:heap-buffer-overflow in logger.c (Reported

     by Badalian Vyacheslav)

* ASTERISK-25685 – infrastructure: Run alembic in Jenkins build

     script (Reported by Joshua Colp)

* ASTERISK-25712 – Second call to already-on-call phone and

     Asterisk sends “Ready” (Reported by Richard Mudgett)

* ASTERISK-24801 – ASAN: ast_el_read_char stack-buffer-overflow

     (Reported by Badalian Vyacheslav)

* ASTERISK-25179 – CDR(billsec,f) and CDR(duration,f) report

     incorrect values (Reported by Gianluca Merlo)

* ASTERISK-25611 – core: threadpool thread_timeout_thrash unit

     test sporadically failing (Reported by Joshua Colp)

* ASTERISK-25686 – PJSIP: qualify_timeout is a double, database

     schema is an integer (Reported by Marcelo Terres)

* ASTERISK-25700 – main/config: Clean config maps on shutdown.

     (Reported by Corey Farrell)

* ASTERISK-25696 – bridge_basic: don’t cache xferfailsound during

     a transfer (Reported by Kevin Harwell)

* ASTERISK-25697 – bridge_basic: don’t play an attended transfer

     fail sound after target hangs up (Reported by Kevin Harwell)

* ASTERISK-25683 – res_ari: Asterisk fails to start if compiled

     with MALLOC_DEBUG  (Reported by yaron nahum)

* ASTERISK-24097 – Documentation – CHANNEL function help text

     missing ‘linkedid’ argument (Reported by Steven T. Wheeler)

* ASTERISK-25690 – Hanging up when executing connected line sub

     does not cause hangup (Reported by Joshua Colp)

* ASTERISK-25687 – res_musiconhold: Concurrent invocations of ‘moh

     reload’ cause a crash (Reported by Sean Bright)

* ASTERISK-25632 – res_pjsip_sdp_rtp: RTP is sent from wrong IP

     address when multihomed (Reported by Olivier Krief)

* ASTERISK-25637 – Multi homed server using wrong IP (Reported by

     Daniel Journo)

* ASTERISK-25394 – pbx: Incorrect device and presence state when

     changing hint details (Reported by Joshua Colp)

* ASTERISK-25640 – pbx: Deadlock on features reload and state

     change hint. (Reported by Krzysztof Trempala)

* ASTERISK-25681 – devicestate: Engine thread is not shut down

     (Reported by Corey Farrell)

* ASTERISK-25680 – manager: manager_channelvars is not cleaned at

     shutdown (Reported by Corey Farrell)

* ASTERISK-25679 – res_calendar leaks scheduler. (Reported by

     Corey Farrell)

* ASTERISK-25675 – Endpoint not listed as Unreachable (Reported by

     Daniel Journo)

* ASTERISK-25677 – pbx_dundi: leaks during failed load. (Reported

     by Corey Farrell)

* ASTERISK-25673 – res_crypto leaks CLI entries (Reported by Corey


* ASTERISK-25668 – res_pjsip: Deadlock in distributor (Reported by

     Mark Michelson)

* ASTERISK-25664 – ast_format_cap_append_by_type leaks a reference

     (Reported by Corey Farrell)

* ASTERISK-25647 – bug of cel_radius.c: wrong point of

     ADD_VENDOR_CODE (Reported by Aaron An)

* ASTERISK-25137 – endpoint stasis messages are delivered twice

     (Reported by Vitezslav Novy)

* ASTERISK-25116 – res_pjsip:  Two PeerStatus AMI messages are

     sent for every status change (Reported by George Joseph)

* ASTERISK-25641 – bridge: GOTO_ON_BLINDXFR doesn’t work on

     transfer initiated channel (Reported by Dmitry Melekhov)

* ASTERISK-25614 – DTLS negotiation delays (Reported by Dade


* ASTERISK-25625 – res_sorcery_memory_cache: Add full backend

     caching (Reported by Joshua Colp)

* ASTERISK-25601 – json: Audit reference usage and thread safety

     (Reported by Joshua Colp)

* ASTERISK-25624 – AMI Event OriginateResponse bug (Reported by

     sungtae kim)

* ASTERISK-25615 – res_pjsip: Setting transport async_operations >

     1 causes segfault on tls transports (Reported by George Joseph)

* ASTERISK-25442 – using realtime (mysql) queue members are never

     updated in wait_our_turn function (app_queue.c)  (Reported by

     Carlos Oliva)

* ASTERISK-25364 – Issue a TCP connection(kernel) and

     thread of asterisk is not released (Reported by Hiroaki Komatsu)

* ASTERISK-25569 – app_meetme: Audio quality issues (Reported by

     Corey Farrell)

* ASTERISK-25619 – res_chan_stats not sending the correct

     information to StatsD (Reported by Tyler Cambron)

* ASTERISK-24146 – No audio on WebRtc caller side when

     answer waiting time is more than ~7sec (Reported by Aleksei


* ASTERISK-25609 – Asterisk may crash when calling

     ast_channel_get_t38_state(c) (Reported by Filip Jenicek)

* ASTERISK-25599 – 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 – format-attribute module: VP8 missing

     (Reported by Alexander Traud)

* ASTERISK-25583 – format-attribute module: RFC 7587 (Opus

     Codec) (Reported by Alexander Traud)

* ASTERISK-25498 – Asterisk crashes when negotiating g729 without

     that module installed (Reported by Ben Langfeld)

* ASTERISK-25595 – Unescaped : in messge sent to statsd (Reported

     by Niklas Larsson)

* ASTERISK-25598 – res_pjsip:  Contact status messages are

     printing a hash instead of the uri (Reported by George Joseph)

* ASTERISK-25600 – bridging: Inconsistency in BRIDGEPEER (Reported

     by Jonathan Rose)

* ASTERISK-25476 – chan_sip loses registrations after a while

     (Reported by Michael Keuter)

* ASTERISK-25593 – fastagi: record file closed after sending

     result (Reported by Kevin Harwell)

* ASTERISK-25585 – 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 – translation module gets cached not

     joint format (Reported by Alexander Traud)

* ASTERISK-25573 – 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


* ASTERISK-25565 – DNS: System resolver only returns 1 record per

     result (Reported by George Joseph)

* ASTERISK-25552 – hashtab: Improve NULL tolerance (Reported by

     Joshua Colp)

* ASTERISK-25160 – Opus Codec: SIP/SDP line fmtp missing

     when called internally (Reported by Alexander Traud)

* ASTERISK-25535 – 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 – format-attribute module: RFC or

     internal defaults? (Reported by Alexander Traud)

* ASTERISK-25533 – 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


* ASTERISK-25528 – DNS: System resolver issues with TTL parse

     (Reported by dtryba)

* ASTERISK-25527 – Quirky xmldoc description wrapping (Reported by

     Walter Doekes)

* ASTERISK-24779 – Passthrough OPUS codec not working with

     chan_pjsip (Reported by PowerPBX)

* ASTERISK-25522 – ARI: Crash when creating channel via ARI

     originate with requesting channel (Reported by Matt Jordan)

* ASTERISK-25434 – Compiler flags not reported in ‘core show

     settings’ despite usage during compilation (Reported by Rusty


* ASTERISK-24106 – WebSockets Automatically decides what driver it

     will use  (Reported by Andrew Nagy)

* ASTERISK-25513 – Crash: malloc failed with high load of

     subscriptions. (Reported by John Bigelow)

* ASTERISK-25505 – res_pjsip_pubsub: Crash on off-nominal when UAS

     dialog can’t be created (Reported by Joshua Colp)

* ASTERISK-25494 – build:  GCC 5.1.x catches some new const, array

     bounds and missing paren issues (Reported by George Joseph)

* ASTERISK-25485 – res_pjsip_outbound_registration: registration

     stops due to 400 response (Reported by Kevin Harwell)

* ASTERISK-25486 – res_pjsip: Fix deadlock when validating URIs

     (Reported by Joshua Colp)

* ASTERISK-7803 – Update the maximum packetization values

     in frame.c (Reported by dea)

* ASTERISK-25484 – autoframing=yes has no effect (Reported

     by Alexander Traud)

* ASTERISK-25308 – ari: Websocket leak (Reported by Joshua Colp)

* ASTERISK-25461 – Nested dialplan #includes don’t work as

     expected. (Reported by Richard Mudgett)

* ASTERISK-25455 – Deadlock of PJSIP realtime over

     res_config_pgsql  (Reported by mdu113)

* ASTERISK-25135 – RTP Timeout hangup cause code missing

     (Reported by Olle Johansson)

* ASTERISK-25108 – configure check for older unbound library

     (Reported by John Bigelow)

* ASTERISK-25435 – Asterisk periodically hangs. UDP Recv-Q greatly

     exceeds zero. (Reported by Dmitriy Serov)

* ASTERISK-25451 – Broken video – erased rtp marker bit (Reported

     by Stefan Engström)

* ASTERISK-25400 – Hints broken when “CustomPresence” doesn’t

     exist in AstDB (Reported by Andrew Nagy)

* ASTERISK-25443 – 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 Nemčić)

* ASTERISK-25441 – Deadlock in res_sorcery_memory_cache. (Reported

     by Richard Mudgett)

* ASTERISK-25438 – res_rtp_asterisk: ICE role message even when

     ICE is not enabled (Reported by Joshua Colp)

* ASTERISK-25383 – Core dumps on startup and shutdown with

     MALLOC_DEBUG enabled (Reported by yaron nahum)

* ASTERISK-25423 – Caller gets no Connected line update during

     call pickup. (Reported by Richard Mudgett)

* ASTERISK-25305 – Dynamic logger channels can be added multiple

     times (Reported by Mark Michelson)

* ASTERISK-25418 – On-hold channels redirected out of a bridge

     appear to still be on hold (Reported by Mark Michelson)

* ASTERISK-25384 – Regular Asterisk crashes when using Page

     application. “user_data is NULL” (Reported by Chet Stevens)

* ASTERISK-25410 – app_record: RECORDED_FILE variable not being

     populated (Reported by Kevin Harwell)

* ASTERISK-25396 – chan_sip: Extremely long callerid name causes

     invalid SIP (Reported by Walter Doekes)

* ASTERISK-25399 – app_queue: AgentComplete event has wrong reason

     (Reported by Kevin Harwell)

* ASTERISK-25185 – Segfault in app_queue on transfer scenarios

     (Reported by Etienne Lessard)

* ASTERISK-25353 – Transcoding while different in Frame

     size = Frames lost (Reported by Alexander Traud)

* ASTERISK-25325 – ARI PUT reload chan_sip HTTP response 404

     (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-25390 – default_from_user can crash with certain

     configuration backends (Reported by Mark Michelson)

* ASTERISK-25387 – res_pjsip_nat: Malformed REGISTER request

     causes NAT’d Contact header to not be rewritten (Reported by

     Matt Jordan)

* ASTERISK-25227 – No audio at in-band announcements in ooh323

     channel (Reported by Alexandr Dranchuk)

* ASTERISK-25295 – res_pjsip crash – pjsip_uri_get_uri at

     /usr/include/pjsip/sip_uri.h (Reported by Dmitriy Serov)

* ASTERISK-25381 – res_pjsip: AoRs deleted via ARI (or other

     mechanism) do not destroy their related contacts (Reported by

     Matt Jordan)

* ASTERISK-25369 – res_parking: ParkAndAnnounce – Inheritable

     variables aren’t applied to the announcer channel (Reported by

     Jonathan Rose)

* ASTERISK-25367 – pbx: Long pattern match hints may cause “core

     show hints” to crash (Reported by Joshua Colp)

* ASTERISK-25365 – Persistent subscriptions have extra

     Content-Length/corrupted messages (Reported by Mark Michelson)

* ASTERISK-25356 – res_pjsip_sdp_rtp: Multiple keepalive scheduled

     items may exist (Reported by Joshua Colp)

* ASTERISK-25355 – sched: ast_sched_del may return prematurely due

     to spurious wakeup (Reported by Joshua Colp)

* ASTERISK-25318 –


     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


* 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


* ASTERISK-25312 – res_http_websocket: Terminate connection on

     fatal cases (Reported by Joshua Colp)

* ASTERISK-25315 – DAHDI channels send shortened duration DTMF

     tones. (Reported by Richard Mudgett)

* ASTERISK-25306 – Persistent subscriptions can save multiple SIP

     messages at once, leading to potential crashes. (Reported by

     Mark Michelson)

* ASTERISK-25309 – iLBC 20 advertised (Reported by

     Alexander Traud)

* ASTERISK-25304 – res_pjsip: XML sanitization may write past

     buffer (Reported by Joshua Colp)

* ASTERISK-25265 – DTLS Failure when calling WebRTC-peer on

     Firefox 39 – add ECDH support and fallback to prime256v1

     (Reported by Stefan Engström)

* ASTERISK-24988 – func_talkdetect: Test is bouncing sporadically

     (Reported by Joshua Colp)

* ASTERISK-25181 – ARI: Channels added to Stasis application

     during WebSocket creation don’t receive a StasisStart event

     (Reported by Matt Jordan)

* ASTERISK-25296 – RTP performance issue with several channel

     drivers. (Reported by Richard Mudgett)

* ASTERISK-25297 – Crashes running

     channels/pjsip/resolver/srv/failover/in_dialog testsuite tests

     (Reported by Richard Mudgett)

* ASTERISK-25292 – Testuite:

     tests/apps/bridge/bridge_wait/bridge_wait_e_options fails

     (Reported by Kevin Harwell)

* ASTERISK-25271 – Parking & blind transfer: Transferer channel

     not hung up if no MOH (Reported by Kevin Harwell)

* ASTERISK-25250 – chan_sip – Despite the channel being answered,

     caller on a call established via Local channel continues to hear

     ringback (Reported by Etienne Lessard)

* ASTERISK-25253 – confbridge volume options and other volume

     controls such as func_volume don’t work (Reported by Dmitriy


* ASTERISK-25247 – choppy audio when spying on a g722 channel,

     chan_sip or chan_pjsip (Reported by hristo)

* ASTERISK-25263 – cdr_adaptive_odbc: CDR insert failure

     due to reversed if logic (Reported by Elazar Broad)

* ASTERISK-24867 – Docs for ‘e’ option in ResetCDR say to use

     CDR_PROP instead, CDR_PROP docs are unclear (Reported by Rusty


* 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 – channels/sig_pri.h -> sig_pri_span ->

     force_restart_unavailable_chans in wrong scope (Reported by

     Patric Marschall)

* ASTERISK-24934 – 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


* ASTERISK-25103 – Roundup – investigate Asterisk DTLS crashes

     (Reported by Rusty Newton)

* ASTERISK-25146 – DNS: Create system level resolver (Reported by

     Joshua Colp)

* ASTERISK-22805 – res_rtp_asterisk: Crash when calling

     BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP

     (Reported by Dmitry Burilov)

* ASTERISK-24550 – res_rtp_asterisk: Crash in

     ast_rtp_on_ice_complete during DTLS handshake (Reported by

     Osaulenko Alexander)

* ASTERISK-24651 – Fix race condition in DTLS (Reported by

     Badalian Vyacheslav)

* ASTERISK-24832 – 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


* ASTERISK-25168 – Random Core Dumps on Asterisk 13.4 PJSIP, in

     ast_channel_name at channel_internal_api.c (Reported by Carl


* ASTERISK-25076 – res_pjsip: Failover does not occur on

     connection-less transport or 503 response (Reported by Joshua


* ASTERISK-25226 – chan_sip: Channel leak in branch 13 on early

     replaces call pickup (Reported by Walter Doekes)

* ASTERISK-25222 – Crash in recurring cancel callback called from

     ast_dns_resolve_cancel on junk pointer (Reported by Matt Jordan)

* ASTERISK-25220 – Closing of fd -1 in chan_mgcp.c

     (Reported by Walter Doekes)

* ASTERISK-25219 – Source and destination overlap in memcpy

     in rtp_engine.c (Reported by Walter Doekes)

* ASTERISK-25212 – Segfault when using DEBUG_FD_LEAKS

     (Reported by Walter Doekes)

* ASTERISK-19277 – 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


* ASTERISK-25189 – AMI: Add Linkedid header to standard channel

     snapshot information. (Reported by Richard Mudgett)

* ASTERISK-25171 – Early completion of feature code attended

     transfer results in intermittent one-way audio, “ghost ringing”

     and robotic sound. (Reported by Rusty Newton)

* ASTERISK-25172 – Crash in channels/sip/sip blind

     transfer/caller_refer_only test in

     ast_format_cap_append_from_cap during ast_request (Reported by

     Matt Jordan)

* ASTERISK-25180 – res_pjsip_mwi: Unsolicited MWI requires reload

     (Reported by Joshua Colp)

* ASTERISK-25182 – 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 – fromtag may need to be updated after

     successful call dialog match (Reported by Damian Ivereigh)

* ASTERISK-25156 – chan_pjsip’s CHAN_START cel event lacks the

     correct context and exten (Reported by cloos)

* ASTERISK-25157 – bridging: Performing a blonde transfer does not

     result in connected line updates (Reported by Joshua Colp)

* ASTERISK-25087 – Asterisk segfault when using Directory

     application with alias option and specific mailbox configuration

     (Reported by Chet Stevens)

* ASTERISK-25115 – Crash related to func

     sip_resolve_invoke_user_callback of res_pjsip/pjsip_resolver.c

     (Reported by John Bigelow)

* ASTERISK-25096 – Segfault when registering over

     websockets with PJSIP (in ast_sockaddr_isnull at

     /include/asterisk/netsock2.h) (Reported by Josh Kitchens)

* ASTERISK-24963 – ASAN: heap-use-after-free with PJSIP and WSS

     (Reported by Badalian Vyacheslav)

* ASTERISK-22559 – gcc 4.6 and higher supports weakref attribute

     but asterisk doesn’t detect it. (Reported by ibercom)

* ASTERISK-25094 – PBX core: Investigate thread safety issues

     (Reported by Corey Farrell)

* ASTERISK-25113 – install_prereq in Debian 8 without “standard

     system utilities” (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-25148 – res_pjsip NULL channel audit (Reported by Mark


* ASTERISK-25131 – chan_pjsip: In-dialog authentication not

     handled. (Reported by Richard Mudgett)

* ASTERISK-24717 – ASAN: global-buffer-overflow codec_{ilbc | gsm

     | adpcm | ipc10} (Reported by Badalian Vyacheslav)

* ASTERISK-25100 – asterisk coredump if host has an IPv6 address

     that end with ::80 (Reported by Mark Petersen)

* ASTERISK-25122 – Large SIP packet received via pjsip over

     websocket crashes Asterisk  (Reported by Ivan Poddubny)

* ASTERISK-25121 – Stasis: Fix unsafe use of stasis_unsubscribe in

     modules. (Reported by Corey Farrell)

* ASTERISK-25120 – Astobj2: Weakproxy subscriptions should be run

     in reverse order. (Reported by Corey Farrell)

* ASTERISK-25105 – res_pjsip:  Possible incompatibility between

     qualify_timeout and pjproject-2.4 (Reported by George Joseph)

* ASTERISK-25117 – res_mwi_external_ami: Fix manager action

     registrations. (Reported by Corey Farrell)

* ASTERISK-25112 – Logger: Configuration settings are not reset to

     default during reload. (Reported by Corey Farrell)

* ASTERISK-24983 – IAX deadlock between hangup and scheduled

     actions (ex. largrq) (Reported by Y Ateya)

* ASTERISK-24944 – main/audiohook.c change prevents G722 call

     recording (Reported by Ronald Raikes)

* ASTERISK-25110 – res_resolver_unbound.c compilation failure:

     SIGURG is undeclared in func unbound_resolver_stop (Reported by

     John Bigelow)

* ASTERISK-24887 – tags in a=crypto lines do not accept 2

     or more digits (Reported by Makoto Dei)

* ASTERISK-25086 – 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


* ASTERISK-25090 – CLI core show channel truncates cdr variables

     (Reported by snuffy)

* ASTERISK-25083 – Message.c: Message channel becomes saturated

     with frames leading to spammy log messages (Reported by Jonathan


* ASTERISK-25085 – Potential crash after unload of

     func_periodic_hook or test_message (Reported by Corey Farrell)

* ASTERISK-25082 – Asterisk deletes message after doing a playback

     of an INBOX message using ast_vm_play when the Old folder is

     full for that mailbox. (Reported by Jonathan Rose)

* ASTERISK-21893 – Segfault after call hangup, in

     ast_channel_hangupcause_set, at channel_internal_api.c (Reported

     by Aleksandr Gordeev)

* ASTERISK-25042 – asterisk.conf options override command-line

     options. (Reported by Corey Farrell)

* ASTERISK-25074 – Regression: Recent clang-related change broke

     cross compiling of Asterisk (Reported by Sebastian Kemper)

* ASTERISK-24442 – Outgoing call files don’t work properly when

     set in the future (Reported by tootai)

* ASTERISK-18252 – queue_log mysql time column data format

     (Reported by Gareth Blades)

* ASTERISK-25041 – Broken column type checking in

     res_config_mysql addon (Reported by Alexandre Fournier)

* ASTERISK-25057 – res_pjsip_pubsub: Crash in send_notify due to

     invalid root pointer in sub_tree (Reported by Matt Jordan)

* ASTERISK-24938 – ARI Snoop Channel results in excessive

     escalating CPU usage (Reported by George Ladoff)

* ASTERISK-25034 – chan_dahdi: Some telco switches occasionally

     ignore ISDN RESTART requests. (Reported by Richard Mudgett)

* ASTERISK-25003 – Asterisk crashes on attended transfer (using

     feature) (Reported by Artem Volodin)

* ASTERISK-25038 – Queue log “EXITWITHTIMEOUT” does not always

     contain waiting time (Reported by Etienne Lessard)

* ASTERISK-25027 – Build System: Many ARI modules are missing

     dependencies. (Reported by Corey Farrell)

* ASTERISK-25061 – pbx_config: Register manager actions with

     module version of macro. (Reported by Corey Farrell)

* ASTERISK-24967 – Problem support schema for pgsql on CEL

     (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-25025 – Periodic crashes (in

     ast_channel_snapshot_create at stasis_channels.c) with Certified

     Asterisk 13. (Reported by Chet Stevens)

* ASTERISK-25053 – Unit test category /main/presence missing

     trailing slash. (Reported by Corey Farrell)

* ASTERISK-22708 – res_odbc.conf negative_connection_cache option

     not respected, failover between DSNs doesn’t work (Reported by


* ASTERISK-25054 – Formats interface’s cannot be unregistered,

     needs to hold modules until shutdown. (Reported by Corey


* ASTERISK-24976 – cdr_odbc not include new columns added on 1.8

     (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-25033 – Asterisk 13 (branch head) won’t compile without

     PJSip (Reported by Peter Whisker)

* ASTERISK-24896 – Using force black background leads to

     colours not being reset (Reported by dant)

* ASTERISK-25048 – Astobj2: Initialization order wrong when both

     refdebug and AO2_DEBUG are both enabled. (Reported by Corey


* ASTERISK-19608 – Asterisk-1.8.x  starts rejecting calls with

     cause code 44 after some time. (Reported by Denis Alberto


* 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


* ASTERISK-24955 – res_fax: v.27ter support baud rate of 2400,

     which is disallowed in res_fax’s check_modem_rate (Reported by

     Matt Jordan)

* ASTERISK-25020 – Mismatched response to outgoing REGISTER

     request (Reported by Mark Michelson)

* ASTERISK-25028 – Build System: Unneeded defines in

     asterisk/buildopts.h (Reported by Corey Farrell)

* ASTERISK-25026 – Git conversion: Non-C files not switched to

     ASTERISK_REGISTER_FILE (Reported by Corey Farrell)

* ASTERISK-24996 – chan_pjsip: Creating Channel Causes Asterisk to

     Crash When Duplicate AOR Sections Exist in pjsip.conf (Reported

     by Ashley Sanders)

* ASTERISK-25018 – pjsip show endpoints crashes asterisk when

     qualified aors present (Reported by Ivan Poddubny)

* ASTERISK-24749 – ConfBridge: Wrong language on playing

     conf-hasjoin and conf-hasleft when played to bridge (Reported by

     Philippe Bolduc)

* ASTERISK-24845 – pjsip send notify not working with Cisco phone

     (Reported by Carl Fortin)

* ASTERISK-25004 – Crash in authenticated reinvite after

     originated T.38 FAX (Reported by Mark Michelson)

* ASTERISK-24999 – PJSIP crashes with malformed contact line

     (Reported by snuffy)

* ASTERISK-24998 – res_corosync:  res_corosync tries to load even

     if res_corosync.conf is missing (Reported by George Joseph)

* ASTERISK-24997 – Astobj2: Some callers of __adjust_lock do not

     pre-check the object (Reported by Corey Farrell)

* ASTERISK-24994 – dns: Query set unit tests are failing due to

     race condition (Reported by Joshua Colp)

* ASTERISK-24982 – res_pjsip_mwi: Unsolicited MWI NOTIFY only sent

     on mailbox changes (Reported by Joshua Colp)

* ASTERISK-24991 – Check for ao2_alloc failure in

     __ast_channel_internal_alloc (Reported by Corey Farrell)

* ASTERISK-24895 – After hangup on the side of the ISDN network no

     HangupRequest event comes for the dahdi channel. (Reported by

     Andrew Zherdin)

* ASTERISK-24977 – Contacts that don’t use qualify are being

     marked as unavailable (Reported by George Joseph)

* ASTERISK-24774 – Segfault in ast_context_destroy with

     extensions.ael and extensions.conf (Reported by Corey Farrell)

* ASTERISK-24841 – ConfBridge: Strange sampling rates chosen when

     channels have multiple native formats (Reported by Matt Jordan)

* ASTERISK-24975 – Enabling ‘DEBUG_THREADLOCALS’ Causes the Build

     to Fail (Reported by Ashley Sanders)

* ASTERISK-24863 – res_pjsip: No endpoint events raised via AMI

     when contacts cannot be reached/qualified (Reported by Dmitriy


* ASTERISK-24869 – Asterisk segfaults on DAHDI attended transfer

     due to application (appl) being NULL on unbridged channel

     (Reported by viniciusfontes)

* ASTERISK-24970 – Crash in res_pjsip_pubsub handling of failed

     notify (Reported by Scott Griepentrog)

* ASTERISK-13271 – menuselect sets defaults too late (Reported by

     John Nemeth)

* ASTERISK-24959 – CLI command cdr show pgsql status

     (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-20524 – AMI improperly handles lines of exactly 1025

     characters (Reported by David M. Lee)

* ASTERISK-24936 – New Feature: AO2 weakproxy objects (Reported by

     Corey Farrell)

* ASTERISK-24954 – Git migration: Asterisk version numbers are

     incompatible with the Test Suite (Reported by Matt Jordan)

* ASTERISK-17608 – cannot be loaded if res_crypto /

     openssl not compiled (Reported by Warren Selby)

* ASTERISK-24928 – 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 – IAX2 custom qualify timer is not taken

     into account (Reported by Frederic Van Espen)

* ASTERISK-24894 – 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


* ASTERISK-24847 – [security] 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 – – 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 – res_pjsip_messaging: Messages may be

     sent out of order (Reported by Mark Michelson)

* ASTERISK-24920 – Asterisk handles duplicate SIP requests as if

     they were each a new request (Reported by Mark Michelson)

* ASTERISK-24781 – PJSIP: Unnecessary 180 Ringing messages sent

     with undesireabe consequences. (Reported by Richard Mudgett)

* ASTERISK-24857 – “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.


* ASTERISK-24155 – 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 – – ASAN: Race condition

     (heap-use-after-free) on asterisk closing (Reported by Badalian


* 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 – Buddies are always auto-registered when

     processing the roster (Reported by Simon Arlott)

* ASTERISK-24780 – – Buddies are always auto-registered

     when processing the roster (Reported by Simon Arlott)

* ASTERISK-24879 – Compilation fails due to 64bit time

     under OpenBSD (Reported by snuffy)

* ASTERISK-24880 – Compilation under OpenBSD  (Reported by


* ASTERISK-21765 – – 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


* 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 – 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 – 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 – – Asterisk terminates when playing a

     voicemail stored in LDAP (Reported by Graham Barnett)

* ASTERISK-24739 – – 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 – 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 – – 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 – Segmentation fault on aoc-e event

     (Reported by Panos Gkikakis)

* ASTERISK-24799 – 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


* ASTERISK-24085 – Documentation – We should remove or further

     document the ‘contact’ section in pjsip.conf (Reported by Rusty


* 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


* 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 – 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


* ASTERISK-24616 – Crash in res_format_attr_h264 due to invalid

     string copy (Reported by Yura Kocyuba)

* ASTERISK-24737 – When agent not logged in, agent status shows

     unavailable, queue status shows agent invalid (Reported by

     Richard Mudgett)

* ASTERISK-24635 – PJSIP outbound PUBLISH crashes when no response

     is ever received (Reported by Marco Paland)

* ASTERISK-24736 – Memory Leak Fixes (Reported by Mark Michelson)

* ASTERISK-24646 – PJSIP changeset 4899 breaks TLS (Reported by

     Stephan Eisvogel)

* ASTERISK-24711 – DTLS handshake broken with latest OpenSSL

     versions (Reported by Jared Biel)

* ASTERISK-24666 – Security Vulnerability: RTP not closed after

     sip call using unsupported codec (Reported by Y Ateya)

* ASTERISK-24676 – Security Vulnerability: URL request injection

     in libCURL (CVE-2014-8150) (Reported by Matt Jordan)

* ASTERISK-24729 – Outbound registration not occuring on new

     registrations after reload. (Reported by Richard Mudgett)

* ASTERISK-24728 – tcptls: Bad file descriptor error when

     reloading chan_sip (Reported by Kevin Harwell)

* ASTERISK-24721 – manager: ModuleLoad action incorrectly reports

     ‘module not found’ during a Reload operation (Reported by Matt


* 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


* 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


* 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 – chan_sip realtime uses case sensitive

     column comparison for ‘defaultuser’ (Reported by


* ASTERISK-24709 – msg_create_from_file used by MixMonitor

     m() option does not queue an MWI event (Reported by Gareth


* ASTERISK-24673 – outgoing sip registers cannot be removed or

     modified without doing restart (or doing module unload (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 – contrib/scripts/install_prereq selects

     32-bit packages on 64-bit hosts (Reported by Ben Klang)

* ASTERISK-24288 – – ODBC usage with app_voicemail –

     voicemail is not deleted after review, hangup (Reported by LEI


* 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 – 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 – 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 – 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 – lacks documentation and does

     not function (Reported by John Kiniston)

* ASTERISK-24637 – Channel re-enters Stasis() when it should not

     (Reported by John Bigelow)

* ASTERISK-24591 – Stasis() side of an ARI originated channel

     cannot be Redirected (Reported by Kinsey Moore)

* ASTERISK-24376 – res_pjsip_refer: REFER request for remote

     session attempts to direct channel to external_replaces

     extension instead of context, without providing for the

     Referred-To SIP URI (Reported by Matt Jordan)

* ASTERISK-24513 – Local channel apparently leaked in off-nominal

     DTMF attended transfer (Reported by Mark Michelson)

* ASTERISK-24367 – PJSIP: allow all results in failure to send

     INVITE (Reported by Scott Griepentrog)

* ASTERISK-24267 – Queue variables associated with

     setinterfacevar, setqueueentryvar, setqueuevar are not passed to

     local channel (Reported by Mitch Claborn)

* ASTERISK-24641 – Deadlock in Trunk (Reported by Malcolm


* ASTERISK-23841 – DTMF atxfer doesn’t set CallerID for the recall

     calls to the transferrer. (Reported by Richard Mudgett)

* ASTERISK-24628 – 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


* 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 – 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


* ASTERISK-24342 – PJSIP: Qualifying endpoints attempts to do them

     all at the same time. (Reported by Richard Mudgett)

* ASTERISK-24556 – Asterisk 13 core dumps when calling from pjsip

     extension to another pjsip extension  (Reported by Abhay Gupta)

* ASTERISK-24537 – Stasis: StasisStart/StasisEnd events are not

     reliably transmitted during transfers (Reported by Matt Jordan)

* ASTERISK-24573 – Out of sync conversation recording when

     divided in multiple recordings (Reported by Nuno Borges)

* ASTERISK-24572 – App_meetme is loaded without its

     defaults when the configuration file is missing (Reported by

     Nuno Borges)

* ASTERISK-24516 – Asterisk segfaults when playing back

     voicemail under high concurrency with an IMAP backend (Reported

     by David Duncan Ross Palmer)

* ASTERISK-24274 – Codec Format Is Not Included in the SDP

     Media Attributes When SLIN48 Codec Is Used (Reported by Frankie


* ASTERISK-24533 – 2 threads created per chan_sip entry (Reported

     by xrobau)

* ASTERISK-24542 – Failure showing codecs via ‘core show

     channeltype <tech>’ (Reported by snuffy)

* ASTERISK-24469 – Security Vulnerability: Mixed IPv4/IPv6 ACLs

     allow blocked addresses through (Reported by Matt Jordan)

* ASTERISK-24534 – Register DB() as escalating to prevent

     users from writing to astdb (Reported by Gareth Palmer)

* ASTERISK-24531 – res_pjsip_acl: ACLs not applied on initial

     module load (Reported by Matt Jordan)

* ASTERISK-24490 – Security Vulnerability: CONFBRIDGE function’s

     record_command option allows arbitrary parameters to be passed

     to MixMonitor, allowing remote execution of commands (Reported

     by Matt Jordan)

* ASTERISK-24528 – res_pjsip_refer: Sending INVITE with Replaces

     in-dialog with invalid target causes crash (Reported by Joshua


* ASTERISK-24471 – Crash – assert_fail in libc in

     pjmedia_sdp_neg_negotiate from /usr/local/lib/

     (Reported by yaron nahum)

* ASTERISK-24535 – stringfields: Fix regression from fix for

     unintentional memory retention and another issue exposed by the

     fix (Reported by Corey Farrell)

* ASTERISK-24508 – pjsip – REFER request from SNOM is rejected

     with “400 bad request” – DEBUG shows “Received a REFER without a

     parseable Refer-To” (Reported by Beppo Mazzucato)

* ASTERISK-15242 – transmit_refer leaks sip_refer structures

     (Reported by David Woolley)

* ASTERISK-24522 – ConfBridge: delay occurs between kicking all

     endmarked users when last marked user leaves (Reported by Matt


* ASTERISK-23651 – Reloading some modules that are loaded already,

     results in ‘No such module’ before a successful reload (Reported

     by Rusty Newton)

* ASTERISK-24336 – PJSIP timer_min_se value under 90 causes crash

     (Reported by Leon Rowland)

* ASTERISK-24501 – ARI: Moving a channel between bridges followed

     by a hangup can cause an ARI client to not receive an expected

     ChannelLeftBridge event before StasisEnd (Reported by Matt


* ASTERISK-24489 – Crash: Asterisk crashes when converting RTCP

     packet to JSON for res_hep_rtcp and report blocks are greater

     than 1 (Reported by Gregory Malsack)

* ASTERISK-24498 – Segmentation fault in res_hep_rtcp on attended

     transfer (Reported by Beppo Mazzucato)

* ASTERISK-24281 – When bridging 2 chan_sip channels, MOH not

     removed from on-hold channels and bridge is never destroyed

     after hangup. (Reported by Stefan Engström)

* ASTERISK-24444 – PBX: Crash when generating extension for

     pattern matching hint (Reported by Leandro Dardini)

* ASTERISK-24502 – Build fails when dev-mode, dont optimize and

     coverage are enabled (Reported by Corey Farrell)

* ASTERISK-24505 – manager: http connections leak references

     (Reported by Corey Farrell)

* ASTERISK-24500 – Regression introduced in chan_mgcp by SVN

     revision r227276 (Reported by Xavier Hienne)

* ASTERISK-24468 – Incoming UCS2 encoded SMS truncated if SMS

     length exceeds 50 (roughly) national symbols (Reported by

     Dmitriy Bubnov)

* ASTERISK-24250 – Voicemail with multi-recipients To:

     header fix (Reported by abelbeck)

* ASTERISK-24504 – chan_console: Fix reference leaks to pvt

     (Reported by Corey Farrell)

* ASTERISK-24447 – Bridge DTMF hooks: Audio doesn’t pass when

     waiting for more matching digits. (Reported by Richard Mudgett)

* ASTERISK-24257 – agent must dial acceptdtmf twice to bridge to

     queue caller (Reported by Steve Pitts)

* ASTERISK-24492 – main/file.c: ast_filestream sometimes causes

     extra calls to ast_module_unref (Reported by Corey Farrell)

* ASTERISK-24491 – Memory leak in res_hep (Reported by Zane


* ASTERISK-24307 – Unintentional memory retention in stringfields

     (Reported by Etienne Lessard)

* ASTERISK-24438 – blocks Asterisk reload

     when DNS settings invalid (Reported by Melissa Shepherd)

* ASTERISK-20127 – [Regression] Config.c config_text_file_load()

     unescapes semicolons (“\;” -> “;”) turning them into comments

     (corruption) on rewrite of a config file (Reported by George


* ASTERISK-24487 – configuration: sections should be loadable as

     template even when not marked (Reported by Scott Griepentrog)

* ASTERISK-24482 – func_talkdetect: Fix stasis message leak in

     audiohook callback (Reported by Corey Farrell)

* ASTERISK-24480 – res_http_websockets: Module reference decrease

     below zero (Reported by Corey Farrell)

* ASTERISK-24476 – main/app.c / app_voicemail: ast_writestream

     leaks (Reported by Corey Farrell)

* ASTERISK-24411 – Status of outbound registration is not

     changed upon unregistering. (Reported by John Bigelow)

* ASTERISK-24432 – Install when REF_DEBUG is enabled

     (Reported by Corey Farrell)

* ASTERISK-24466 – app_queue: fix a couple leaks to struct

     call_queue (Reported by Corey Farrell)

* ASTERISK-24465 – audiohooks list leaks reference to formats

     (Reported by Corey Farrell)

* ASTERISK-24462 – res_pjsip: Stale qualify statistics after

     disablementation (Reported by Kevin Harwell)

* ASTERISK-24190 – IMAP voicemail causes segfault (Reported by

     Nick Adams)

* ASTERISK-24304 – asterisk crashing randomly because of unistim

     channel (Reported by dhanapathy sathya)

* ASTERISK-21721 – SIP Failed to parse multiple Supported: headers

     (Reported by Olle Johansson)

* ASTERISK-24458 – chan_phone fails to build on big endian systems

     (Reported by Tzafrir Cohen)

* ASTERISK-24457 – res_fax: fax gateway frames leak (Reported by

     Corey Farrell)

* ASTERISK-24453 – manager: acl_change_sub leaks (Reported by

     Corey Farrell)

* ASTERISK-24437 – Review implementation of ast_bridge_impart for

     leaks and document proper usage (Reported by Scott Griepentrog)

* ASTERISK-24430 – missing letter “p” in word response in

     OriginateResponse event documentation (Reported by Dafi Ni)

* ASTERISK-24323 – Bug in documentation AGI STREAM FILE CONTROL

     (Reported by Martin Cisárik)

* ASTERISK-24419 – Incorrect syntax for setting language in

     configs/extensions.conf.sample (Reported by Ben Klang)

* ASTERISK-24454 – app_queue: ao2_iterator not destroyed, causing

     leak (Reported by Corey Farrell)

* ASTERISK-24455 – func_cdr: CDR_PROP leaks payload (Reported by

     Corey Farrell)

* ASTERISK-24435 – Asterisk 13 with TC400P segfault (Reported by

     Marian Koniuszko)

* ASTERISK-24425 – jabber/xmpp to use TLS instead of

     SSLv3, security fix POODLE (CVE-2014-3566) (Reported by


* ASTERISK-24122 – Documentaton for res_pjsip option use_avpf

     needs to be fixed (Reported by James Van Vleet)

* ASTERISK-24381 – res_pjsip_sdp_rtp: Declined media streams are

     interpreted, leading to erroneous 488 rejections (Reported by

     Matt Jordan)

* ASTERISK-24063 – Asterisk does not respect outbound proxy

     when sending qualify requests (Reported by Damian Ivereigh)

* ASTERISK-24415 – Missing AMI VarSet events when channels inherit

     variables. (Reported by Richard Mudgett)

* ASTERISK-24327 – bridge_native_rtp: Smart bridge operation to

     softmix sometimes fails to properly re-INVITE remotely bridged

     participants (Reported by Matt Jordan)

* ASTERISK-24426 – CDR Batch mode: size used as time value after

     first expire (Reported by Shane Blaser)

* ASTERISK-24312 – SIGABRT when improperly configured realtime

     pjsip  (Reported by Dafi Ni)

* ASTERISK-23846 – Unistim multilines. Loss of voice after second

     call drops (on a second line). (Reported by Rustam Khankishyiev)

* ASTERISK-24413 – parking/parking_tests: Crash due to assertion

     in unit tests when MoH is started on channel in holding bridge

     (Reported by Matt Jordan)

* ASTERISK-24393 – rtptimeout=0 doesn’t disable rtptimeout

     (Reported by Dmitry Melekhov)

* ASTERISK-24321 – SIP deadlock when running automated queues

     tests (Reported by Steve Pitts)

* ASTERISK-24392 – res_fax: fax gateway sessions leak (Reported by

     Corey Farrell)

* ASTERISK-24237 – CDR: FRACK With PJSIP blonde transfer.

     (Reported by Richard Mudgett)

* ASTERISK-24394 – CDR: FRACK with PJSIP directed pickup.

     (Reported by Richard Mudgett)

* ASTERISK-18923 – res_fax_spandsp usage counter is wrong

     (Reported by Grigoriy Puzankin)

* ASTERISK-22791 – asterisk sends Re-INVITE after receiving a BYE

     (Reported by not here)

* ASTERISK-13797 – relax badshell tilde test (Reported by

     Tzafrir Cohen)

* ASTERISK-24325 – res_calendar_ews: cannot be used with neon 0.30

     (Reported by Tzafrir Cohen)

* ASTERISK-24406 – Some caller ID strings are parsed differently

     since 11.13.0 (Reported by Etienne Lessard)

* ASTERISK-24387 – res_pjsip: rport sent from UAS MUST include the

     port that the UAC sent the request on (Reported by Matt Jordan)

* ASTERISK-20784 – Failure to receive an ACK to a SIP Re-INVITE

     results in a SIP channel leak (Reported by NITESH BANSAL)

* ASTERISK-15879 – Failure to receive an ACK to a SIP

     Re-INVITE results in a SIP channel leak (Reported by Torrey


* ASTERISK-24383 – res_rtp_asterisk: Crash if no candidates

     received for component (Reported by Kevin Harwell)

* ASTERISK-24011 – safe_asterisk tries to set ulimit -n too

     high on linux systems with lots of RAM (Reported by Michael


* ASTERISK-24326 – res_rtp_asterisk: ICE-TCP candidates are

     incorrectly attempted (Reported by Joshua Colp)

* ASTERISK-24389 – chan_iax2: Unit test on Bamboo failing

     (Reported by Kevin Harwell)

* ASTERISK-24398 – Initialize auth_rejection_permanent on client

     state to the configuration parameter value (Reported by Matt


* ASTERISK-24354 – AMI sendMessage closes AMI connection on error

     (Reported by Peter Katzmann)

* ASTERISK-24224 – When using Bridge() dialplan application,

     surrogate channel appears in list and call count is inflated.

     (Reported by Mark Michelson)

* ASTERISK-24370 – res_pjsip/pjsip_options: OPTIONS request sent

     to Asterisk with no user in request is always 404’d (Reported by

     Matt Jordan)

* ASTERISK-24382 – chan_pjsip: Calling PJSIP_MEDIA_OFFER on a

     non-PJSIP channel results in an invalid reference of a channel

     pvt and a FRACK (Reported by Matt Jordan)

* ASTERISK-24369 – res_pjsip: Large message on reliable transport

     can cause empty messages to be passed from the PJSIP stack up,

     causing crashes in multiple locations (Reported by Matt Jordan)

* ASTERISK-24368 – res_pjsip_pubsub: Subscription persistence

     causes crash when re-constructing stored subscription (Reported

     by Matt Jordan)

* ASTERISK-24378 – Release AMI connections on shutdown (Reported

     by Corey Farrell)

* ASTERISK-24384 – chan_motif: format capabilities leak on module

     load error (Reported by Corey Farrell)

* ASTERISK-24199 – ‘ALL’ is specified in pjsip.conf.sample for TLS

     cipher but it is not valid (Reported by Joshua Colp)

* ASTERISK-24195 – bridge_native_rtp: Removing mixmonitor from a

     native RTP capable smart bridge doesn’t cause the bridge to

     resume being a native rtp bridge (Reported by Jonathan Rose)

* ASTERISK-24356 – PJSIP: Directed pickup causes deadlock

     (Reported by Richard Mudgett)

* ASTERISK-24262 – AMI CoreShowChannel missing several output

     fields and event documentation (Reported by Mitch Claborn)

* ASTERISK-23781 – outgoing missing as enum from

     contrib/ast-db-manage/config (Reported by Stephen More)

* ASTERISK-24222 – PJSIP: Failed assertions when placing a call

     with no allow= specified (Reported by Mark Michelson)

* ASTERISK-24362 – res_hep leaks reference to configuration

     (Reported by Corey Farrell)

* ASTERISK-22945 – Memory leaks in chan_sip.c with

     realtime peers (Reported by ibercom)

* ASTERISK-24350 – PJSIP shows commands prints unneeded headers

     (Reported by snuffy)

* ASTERISK-20567 – bashism in autosupport (Reported by Tzafrir


* ASTERISK-24357 – [fax] Out of bounds error in update_modem_bits

     (Reported by Jeremy Lainé)

* ASTERISK-24348 – Built-in editline tab complete segfault with

     MALLOC_DEBUG (Reported by Walter Doekes)

* ASTERISK-23768 – Asterisk man page contains a (new)

     unquoted minus sign (Reported by Jeremy Lainé)

* ASTERISK-24295 – crash: creating out of dialog OPTIONS request

     crashes (Reported by Rogger Padilla)

* ASTERISK-24335 – [PATCH] Asterisk incorrectly responds 503 to

     INVITE retransmissions of rejected calls (Reported by Torrey


* ASTERISK-24339 – Swagger API Docs have incorrect basePath

     (Reported by Bradley Watkins)

* ASTERISK-24265 – segfault in asterisk when try to make call to

     IAX  (Reported by Dafi Ni)

* ASTERISK-24290 – Endpoint identifier match value fails to parse

     when CIDR network format is specified (Reported by Ray Crumrine)

* ASTERISK-24301 – Security: Out of call MESSAGE requests

     processed via Message channel driver can crash Asterisk

     (Reported by Matt Jordan)

* ASTERISK-24136 – Security: Crash in Asterisk’s PJSIP code when

     subscribing to an event with an unexpected body type (Reported

     by Mark Michelson)

* ASTERISK-24161 – PJSIPShowEndpoint gives inaccurate count of

     list items (Reported by Mark Michelson)

* ASTERISK-24331 – Unexpected Errors in Asterisk Manager Interface

     Output (Reported by xrobau)

* ASTERISK-24328 – Use of MixMonitor ‘m’ option results in 0

     duration vm description file  (Reported by Scott Griepentrog)

* ASTERISK-23577 – res_rtp_asterisk: Crash in

     ast_rtp_on_turn_rtp_state when RTP instance is NULL (Reported by

     Jay Jideliov)

* ASTERISK-23634 – With TURN Asterisk crashes on multiple (7-10)

     concurrent WebRTC (avpg/encryption/icesupport) calls (Reported

     by Roman Skvirsky)

* ASTERISK-24249 – SIP debugs do not stop (Reported by Avinash


* ASTERISK-24181 – RLS: Large lists don’t get sent because they

     exceed the PJSIP message length limit (Reported by Jonathan


* ASTERISK-24254 – CDRs: Application/args/dialplan CEP updated

     during dial operation (Reported by Matt Jordan)

* ASTERISK-24241 – crash: CDRs recursively attempt to update Party

     B information in a multi-party bridge, overrunning the stack

     (Reported by Deepak Singh Rawat)

* ASTERISK-24208 – Channels with CDR Information Remain Active

     Even After ConfBrige Is Ended (Reported by Frankie Chin)

* ASTERISK-24223 – Gibberish Call-ID on Local channel on

     origination (Reported by Mark Michelson)

* ASTERISK-24271 – Unable to make WebRTC call through chan_PJSIP

     nor chan_SIP (Reported by Dafi Ni)

* ASTERISK-24212 – testsuite: Sporadic crash due to assert on

     stopping RTP engine (Reported by Matt Jordan)

* ASTERISK-24264 – ARI: Adding a channel to a holding bridge

     automatically starts MOH (Reported by Samuel Galarneau)

* ASTERISK-23767 – Dynamic IAX2 registration stops trying

     if ever not able to resolve (Reported by David Herselman)

* ASTERISK-24280 – Add ‘rtpbindaddr’ setting for chan_sip

     (Reported by Paul Belanger)

* ASTERISK-24019 – When a Music On Hold stream starts it restarts

     at beginning of file. (Reported by Jason Richards)

* ASTERISK-24143 – pjsip: Outbound call to WebRTC UA fails to

     transmit ACK on received 200 OK (Reported by Aleksei Kulakov)

* ASTERISK-23997 – chan_sip: port incorrectly incremented for RTCP

     ICE candidates in SDP answer (Reported by Badalian Vyacheslav)

* ASTERISK-24147 – ARI: channel hangup crashes asterisk process

     (Reported by Edvin Vidmar)

* ASTERISK-23994 – res_pjsip_sdp_rtp: owner address in SDP may not

     be fully qualified domainname (Reported by Private Name)

* ASTERISK-22252 – res_musiconhold cleanup – REF_DEBUG reload

     warnings and ref leaks (Reported by Walter Doekes)

* ASTERISK-24178 – fromdomainport used even if not set

     (Reported by Elazar Broad)

* ASTERISK-24229 – ARI: playback of sounds implicitly answers

     channel, preventing early media playback (Reported by Matt


* ASTERISK-24245 – gcc 4.1.2 complains of files that do not end

     with newlines (Reported by Shaun Ruffell)

* ASTERISK-24246 – Quiet warning about type qualifiers ignored on

     function return type (Reported by Shaun Ruffell)

* ASTERISK-24043 – ARI /continue fails to actually continue into

     the dialplan (Reported by Krandon Bruse)

* ASTERISK-24215 – testsuite: ARI Live Dangerously test fails due

     to wrong response code from Asterisk (Reported by Matt Jordan)

* ASTERISK-24134 – ARI: GET /channels/{channel_id}/variable for

     channel in dialplan returns 409 conflict (Reported by Matt


* ASTERISK-24138 – dial: Call forwarding information presented

     through AMI/ARI is wrong (Reported by Matt Jordan)

* ASTERISK-24234 – app_meetme: Crash on conference shutdown due to

     NULL channel passed to meetme_stasis_generate_msg() (Reported by

     Shaun Ruffell)

* ASTERISK-24225 – Dial option z is broken (Reported by


* ASTERISK-24032 – Gentoo compilation emits warning:

     “_FORTIFY_SOURCE” redefined (Reported by Kilburn)

* ASTERISK-24027 – MixMonitor AMI action called during AGI

     execution from bridge feature causes channel to leave AGI has

     hung up (Reported by Matt Jordan)

* ASTERISK-24236 – res_hep_rtcp: Module incorrectly depends on

     pjsip (Reported by Matt Jordan)

* ASTERISK-23508 – Memory Corruption in

     __ast_string_field_ptr_build_va (Reported by Arnd Schmitter)

Improvements made in this release:


* ASTERISK-26218 – iLBC 20 (Reported by Alexander Traud)

* ASTERISK-26190 – SRTP: Enable AES-256 and AES-GCM.

     (Reported by Alexander Traud)

* ASTERISK-26220 – Add support for noreturn function attributes.

     (Reported by Corey Farrell)

* ASTERISK-22131 – Update the make dependencies script to pull,

     build, and install the correct pjproject (Reported by Matt


* ASTERISK-25471 – Add subscribe_context to res_pjsip

     (Reported by JoshE)

* ASTERISK-26159 – res_hep: enabled by default and information

     sent to default address (Reported by Ross Beer)

* ASTERISK-26088 – Investigate heavy memory utilization by

     res_pjsip_pubsub (Reported by Richard Mudgett)

* ASTERISK-25578 – SIP/SDP: No rtpmap for static RTP

     payload IDs (Reported by Alexander Traud)

* ASTERISK-26011 – PJSIP: add “via_addr”, “via_port”,

     “call_id” to contacts (Reported by Alexei Gradinari)

* ASTERISK-25965 – res_pjsip_outbound_publish: Allow multiple

     clients per configuration (Reported by Kevin Harwell)

* ASTERISK-25994 – res_pjsip: module load priority

     (Reported by Alexei Gradinari)

* ASTERISK-25931 – PJSIP: add “reg_server” to contacts. (Reported

     by Alexei Gradinari)

* ASTERISK-25835 – Authentication using ‘Username’ field from

     Digest (Reported by Ross Beer)

* ASTERISK-25930 – PJSIP: disable multi domain to improve realtime

     performace (Reported by Alexei Gradinari)

* ASTERISK-25865 – Message-Account Missing From PJSIP MWI

     (Reported by Ross Beer)

* ASTERISK-25444 – Music On Hold Warning misleading

     (Reported by Conrad de Wet)

* ASTERISK-25846 – Gracefully deal with Absent Stasis Apps

     (Reported by Andrew Nagy)

* ASTERISK-25791 – res_pjsip_caller_id: Lack of support for

     Anonymous <anonymous@anonymous.invalid> (Reported by Anthony


* ASTERISK-25767 – Add check to configure for sanitizes

     (Reported by Badalian Vyacheslav)

* ASTERISK-25068 – Move commonly used FreePBX extra sounds to the

     core set (Reported by Rusty Newton)

* ASTERISK-25627 – Easily Preventable Compile Warning (Reported by

     Diederik de Groot)

* ASTERISK-25558 – chan_sip option ‘notifyringing’ doc fix

     and addition of ‘notifyringingprio’ (Reported by Ward van


* ASTERISK-25618 – res_pjsip:  Check for readability of TLS files

     at startup (Reported by George Joseph)

* ASTERISK-25581 – Add value reason a pause on CLI

     (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-25572 – Endpoints: Add StatsD stats for Asterisk

     endpoints (Reported by Matt Jordan)

* ASTERISK-25571 – PJSIP: Add StatsD stats for some common PJSIP

     objects (Reported by Matt Jordan)

* ASTERISK-25518 – taskprocessor: Add high water mark (Reported by

     Jonathan Rose)

* ASTERISK-25495 – Prevent old-update packages on

     repository Debian systems (Reported by Rodrigo Ramirez


* ASTERISK-25477 – pjsip show “command” like [criteria] (Reported

     by Bryant Zimmerman)

* ASTERISK-24718 – Add inital support of “sanitize” to

     configure (Reported by Badalian Vyacheslav)

* ASTERISK-24870 – ARI: Subscriptions to bridges generally not

     super useful (Reported by Matt Jordan)

* ASTERISK-25405 – CLI: core show fd: add timestamp

     (Reported by Alexander Traud)

* ASTERISK-25310 – on FreeBSD also pthread_attr_init()

     defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)

* ASTERISK-25256 – Post AMI VarSet to empty string events

     when Asterisk deletes a dialplan variable. (Reported by Richard


* ASTERISK-25040 – pbx: Improve performance of reloads by making

     hint destruction more performant (Reported by Matt Jordan)

* ASTERISK-25067 – Sorcery Caching: Implement a new caching module

     (Reported by Matt Jordan)

* ASTERISK-25114 – res_pjsip:  Add AMI events for chan_pjsip

     contact lifecycle changes (Reported by George Joseph)

* ASTERISK-25072 – res_pjsip_outbound_registration: line

     functionality. Additional check for using the request URI

     (Reported by Dmitriy Serov)

* ASTERISK-24815 – Enable TLS Dual-Certificates (ECC+RSA)

     (Reported by Alexander Traud)

* ASTERISK-25063 – add X.509 subject alternative name

     support to Asterisk TLS support (Reported by Maciej Szmigiero)

* ASTERISK-25044 – sorcery:  Add ability to insert a new wizard

     into an object type’s list (Reported by George Joseph)

* ASTERISK-24892 – Super Awesome Company sound prompts (Reported

     by Rusty Newton)

* ASTERISK-24744 – Swedish Core Voice prompts (Reported by Tove


* ASTERISK-25049 – CLI: Enable automatic references to modules

     (Reported by Corey Farrell)

* ASTERISK-25056 – Modules: Make ast_module_info->self available

     to auxiliary sources.  (Reported by Corey Farrell)

* ASTERISK-25045 – vector:  Add new capabilities and unit tests

     (Reported by George Joseph)

* ASTERISK-25043 – Avoiding ERR_remove_state in OpenSSL

     (Reported by Alexander Traud)

* ASTERISK-24706 – add auto-dtmf mode for pjsip (Reported

     by yaron nahum)

* ASTERISK-24917 – clang compilation warnings (Reported by

     Diederik de Groot)

* ASTERISK-25051 – Remove unneeded uses of optional_api providers.

     (Reported by Corey Farrell)

* ASTERISK-24974 – Astobj2: Allow reference debugging to be

     enabled/disabled by config. (Reported by Corey Farrell)

* ASTERISK-24980 – cdr_adaptive_odbc: refactor lines to

     concatenate  of columns name (Reported by Rodrigo Ramirez


* ASTERISK-24947 – res_pjsip: Add a PJSIP resolver using core DNS

     (Reported by Joshua Colp)

* ASTERISK-24965 – cel_pgsql – log_error string references CDR

     instead of CEL (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-24960 – Build System: Create MOD_ADD_SOURCE macro for

     module Makefiles (Reported by Corey Farrell)

* ASTERISK-24939 – IAX make calltoken expiration time

     configurable (Reported by Y Ateya)

* ASTERISK-24918 – pjsip: add CLI options to display global and

     system configuration (Reported by Scott Griepentrog)

* ASTERISK-24862 – 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 – Please support Clang; Allow no-exec

     stacks (Reported by Jeffrey Walton)

* ASTERISK-24790 – Reduce spurious noise in logs from voicemail –

     Couldn’t find mailbox %s in context (Reported by Graham Barnett)

* ASTERISK-24813 – asterisk.c: #if statement in listener()

     confuses code folding editors (Reported by Corey Farrell)

* ASTERISK-24811 – asterisk-publication sorcery object does not

     use realtime (Reported by Matt Hoskins)

* ASTERISK-24745 – 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 – 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 – Incomplete channel originate/continue

     handling with ARI (Reported by Nir Simionovich (GreenfieldTech –


* ASTERISK-24351 – Allow passing options and command to

     MixMonitor when recording in ConfBridge (Reported by Gareth


* ASTERISK-24553 – ARI/AMI: Include language in standard channel

     snapshot output (Reported by Matt Jordan)

* ASTERISK-24552 – ARI: Allow associating a channel as an

     initiator of an Origination for record keeping purposes

     (Reported by Matt Jordan)

* ASTERISK-24577 – Speed up loopback switches by avoiding unneeded

     lookups (Reported by Birger “WIMPy” Harzenetter)

* ASTERISK-24530 – app_record stripping 1/4 second from

     recordings (Reported by Ben Smithurst)

* ASTERISK-24283 – Microseconds precision in the eventtime

     column in the cel_odbc module (Reported by Etienne Lessard)

* ASTERISK-24128 – [Patch] Adding default dtls settings (Reported

     by Michael K.)

* ASTERISK-24279 – Documentation: Clarify the behaviour of the CDR

     property ‘unanswered’ (Reported by Matt Jordan)

* ASTERISK-23512 – Inaccurate comment in manager.conf.sample

     (Reported by Richard Miller)

* ASTERISK-24365 – [Patch] Dialplan function to get first/head

     caller channel on queue (Reported by Kristian Høgh)

* ASTERISK-23324 – – QLOOG commiting Japanese translated

     prompts (Reported by Kevin McCoy)

* ASTERISK-24038 – device state: Report ONHOLD device state if

     channel driver defers device state calculation to core (Reported

     by Matt Jordan)

* ASTERISK-24171 – Provide a manpage for the aelparse

     utility (Reported by Jeremy Lainé)

* ASTERISK-23953 – Testsuite: Off-nominal Authenticate test

     (Reported by Matt Jordan)

* ASTERISK-24045 – Voicemail to email at multiple email

     addresses (Reported by Jacob Barber)

For a full list of changes in this beta, please see the ChangeLog:

Thank you for your continued support of Asterisk!


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