[asterisk-users] Asterisk 14.0.0-beta1 Now Available

Asterisk Development Team asteriskteam at digium.com
Wed Jul 27 16:02:35 CDT 2016

The Asterisk Development Team has announced the first beta of
Asterisk 14.0.0. This beta is available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk

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

The following are the issues resolved in this beta:

New Features made in this release:
 * ASTERISK-25904 - PJSIP: add contact.updated event (Reported by
      Alexei Gradinari)
 * ASTERISK-26058 - [Patch] Add uptime and last reloaded to
      FullyBooted AMI event (Reported by Niklas Larsson)
 * ASTERISK-25925 - Allow Early Bridges on ARI Dials (Reported by
      Mark Michelson)
 * ASTERISK-26068 - Multicast RTP Options (Reported by Mark
 * ASTERISK-26042 - ARI: Allow downloading of the media associated
      with a stored recording (Reported by Matt Jordan)
 * ASTERISK-25425 - logger: Add JSON structured logging (Reported
      by Matt Jordan)
 * ASTERISK-25900 - PJSIP Endpoint IP Access Controls (Reported by
      Alexei Gradinari)
 * ASTERISK-25972 - res_pjsip_exten_state: Use body generator to
      publish extension state (Reported by Richard Mudgett)
 * ASTERISK-25889 - ARI: Add separate "create" and "dial"
      operations for channels (Reported by Mark Michelson)
 * ASTERISK-25803 - [patch] chan_sip: Optionally supply
      fromuser/fromdomain in SIP dial string (Reported by Walter
 * 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 spandspflow2pcap.py to
      contrib/scripts. (Reported by Walter Doekes)
 * ASTERISK-25591 - [patch] Complete List of Header Files
      (#include): iwyu (Reported by Alexander Traud)
 * ASTERISK-25551 - [patch]Ability to add channel to an existing
      bridge by specifying an existing channel prefix (Reported by
      Alec Davis)
 * ASTERISK-25419 - Dialplan Application for Integration of StatsD
      (Reported by Ashley Sanders)
 * ASTERISK-25549 - Confbridge: Add participant timeout option
      (Reported by Mark Michelson)
 * ASTERISK-24922 - ARI: Add the ability to intercept hold and
      raise an event (Reported by Matt Jordan)
 * ASTERISK-25479 - Allow CDR's to be modified before being
      dispatched to engines (Reported by Jonh Wendell)
 * ASTERISK-25480 - [patch]Add field PauseReason on
      QueueMemberStatus (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-25377 - res_pjsip: Change default "From user" from UUID
      to something more palatable (Reported by Mark Michelson)
 * ASTERISK-25252 - ARI: Add the ability to manipulate log channels
      (Reported by Matt Jordan)
 * ASTERISK-25259 - chan_pjsip: Add rtptimeout support (Reported by
      Joshua Colp)
 * ASTERISK-25238 - ARI: Support push configuration (Reported by
      Matt Jordan)
 * ASTERISK-25173 - ARI: Add the ability to load/reload/unload an
      Asterisk module (Reported by Matt Jordan)
 * ASTERISK-25006 - [patch] Add support set character for quoted
      identifiers  (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-23186 - [patch] Add usegmtime option to cel_pgsql
      (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-24931 - dns: Add support for SRV records. (Reported by
      Joshua Colp)
 * ASTERISK-24834 - DNS Overhaul: Implement the proposed core API -
      sync/async functions, resolver registration (Reported by Matt
 * ASTERISK-24836 - DNS Overhaul: Write a Resolver Implementation
      (Reported by Matt Jordan)
 * ASTERISK-22591 - [patch]Prevent Asterisk from writing received
      SMS content in log (Reported by Jan Juergens)
 * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
      (Reported by Dwayne Hubbard)
 * ASTERISK-24703 - ARI: Add the ability to "transfer" (redirect) a
      channel (Reported by Matt Jordan)
 * ASTERISK-24363 - [patch] Add ability for Channel Drivers to
      provide Presence State information (Reported by Gareth Palmer)
 * ASTERISK-24554 - AMI/ARI: Generate events on connected line
      changes (Reported by Matt Jordan)
 * ASTERISK-24276 - [Patch] Option to make app MOH override channel
      musicclass (Reported by Kristian Høgh)
 * ASTERISK-23871 - RLS Tests: Implement RLS off-nominal tests
      (Reported by Mark Michelson)
 * ASTERISK-23823 - [patch] Option to keep queuerules in realtime
      (Reported by Michael K.)

Bugs fixed in this release:
 * ASTERISK-26227 - sqlalchemy error due to long identifier name
      (Reported by Mark Michelson)
 * ASTERISK-26221 - chan_sip: iLBC does not include correct mode
      (Reported by Aaron Meriwether)
 * ASTERISK-26216 - res_fax: Deadlock when detect fax while channel
      executing Playback (Reported by Richard Mudgett)
 * ASTERISK-26214 - Allow arbitrary time for fax detection to end
      on a channel (Reported by Richard Mudgett)
 * ASTERISK-23013 - [patch] Deadlock between 'sip show channels'
      command and attended transfer handling (Reported by Ben
 * ASTERISK-26212 - [patch] Makefile: Retain XML Declaration and
      DTD in docs. (Reported by Alexander Traud)
 * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in
      conditional code. (Reported by Corey Farrell)
 * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence
      number even on lost packets. (Reported by Alexander Traud)
 * ASTERISK-26038 - 'make install' doesn't seem to install OS/X
      init files (Reported by Tzafrir Cohen)
 * ASTERISK-26133 - app_queue: Queue members receive multiple calls
      (Reported by Richard Miller)
 * ASTERISK-26196 - pbx: Time based includes can leak timezone
      string (Reported by Corey Farrell)
 * ASTERISK-26193 - chan_sip: reference leak in mwi_event_cb
      (Reported by Corey Farrell)
 * ASTERISK-26191 - threadpool: Leak on duplicate taskprocessor for
      ast_threadpool_serializer_group (Reported by Corey Farrell)
 * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing
      DTLS failure occurred on RTP instance (Reported by Edwin
 * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf.
      (Reported by Alexander Traud)
 * ASTERISK-26160 - pjsip: Updated->Reachable during qualify
      (Reported by Matt Jordan)
 * ASTERISK-26177 - func_odbc: Database handle is kept when it
      should be released (Reported by Leandro Dardini)
 * ASTERISK-25289 - Build System does not respect CFLAGS and
      CXXFLAGS when building menuselect (Reported by Jeffrey Walton)
 * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out
      of bounds and bugs (Reported by Alexei Gradinari)
 * ASTERISK-26184 - chan_sip: Reference leaks in error paths.
      (Reported by Corey Farrell)
 * ASTERISK-26181 - REF_DEBUG: Node object incorrectly logged
      during duplicate replacement (Reported by Corey Farrell)
 * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported
      by Joshua Colp)
 * ASTERISK-26180 - PJSIP: provide valid tcp nodelay option for
      reuse (Reported by Scott Griepentrog)
 * ASTERISK-25772 - res_pjsip: Unexpected two BYE when answered
      (Reported by Dmitriy Serov)
 * ASTERISK-26099 - res_pjsip_pubsub: Crash when sending request
      due to server timeout (Reported by Ross Beer)
 * ASTERISK-26144 - Crash on loading codecs g729/g723 (Reported by
      Alexei Gradinari)
 * ASTERISK-26157 - Build:   Fix errors highlighted by GCC 6.x
      (Reported by George Joseph)
 * ASTERISK-26021 - Build codecs siren7 and siren14 for Asterisk 13
      (Reported by Daniel Denson)
 * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to
      v21_details (Reported by Corey Farrell)
 * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a
      self-comparison (Reported by George Joseph)
 * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim
      generates a compile error (Reported by George Joseph)
 * ASTERISK-26128 - Alembic scripts are failing (Reported by Mark
 * ASTERISK-26139 - test_res_pjsip_scheduler:  Compile failure if
      pjproject isn't installed in a system location (Reported by
      George Joseph)
 * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version.
      (Reported by Alexander Traud)
 * ASTERISK-26132 - PJSIP: provide transport type with received
      messages (Reported by Scott Griepentrog)
 * ASTERISK-26127 - res_pjsip_session: Crash due to race condition
      between res_pjsip_session unload and timer (Reported by Joshua
 * ASTERISK-26045 - [patch]app_voicemail: fix bugs, imap mm_status
      log change to debug (Reported by Alexei Gradinari)
 * ASTERISK-26083 - ARI: Announcer channels staying around after
      playback to a bridge is finished (Reported by Per Jensen)
 * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in
      http.conf (Reported by Alexander Traud)
 * ASTERISK-26097 - [patch] CLI: show maximum file descriptors
      (Reported by Alexander Traud)
 * ASTERISK-25262 - Memory leak when a caller channel does multiple
      dials and CEL is enabled (Reported by Etienne Lessard)
 * ASTERISK-26047 - ARI allows certain commands to run on down
      channels. (Reported by Mark Michelson)
 * ASTERISK-25959 -
      http_media_cache/retrieve_cache_control_directives: Sporadic
      failure (Reported by Joshua Colp)
 * ASTERISK-26103 - cdr:  Assert on 'dial end' event during a blond
      transfer (Reported by George Joseph)
 * ASTERISK-26092 - [Segfault] in res_rtp_asterisk.c:4268 after
      Remotely bridged channels (Reported by Niklas Larsson)
 * ASTERISK-26089 - Invalid security events during boot using PJSIP
      Realtime (Reported by Scott Griepentrog)
 * ASTERISK-26096 - res_hep: Crash when configuration file is
      missing (Reported by Niklas Larsson)
 * ASTERISK-26074 - res_odbc: Deadlock within UnixODBC (Reported by
      Ross Beer)
 * ASTERISK-26054 - Asterisk crashes (core dump) (Reported by B.
 * ASTERISK-26069 - Asterisk truncates To: header, dropping the
      closing '>' (Reported by Vasil Kolev)
 * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling
      against libsrtp-1.5.0 (Reported by Patrick Laimbock)
 * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar
      cr (Reported by Alexander Traud)
 * ASTERISK-26070 - ari/channels:  Creating a local channel without
      an originator adds all audio formats to it's capabilities
      (Reported by George Joseph)
 * ASTERISK-26078 - core: Memory leak in logging (Reported by
      Etienne Lessard)
 * ASTERISK-26065 - chan_pjsip: MWI NOTIFY contents not ordered
      properly (Reported by Ross Beer)
 * ASTERISK-26063 - ${PJSIP_HEADER(read,Call-ID)} does not work -
      documentation needs clarification for when read/write is
      possible (Reported by Private Name)
 * ASTERISK-25777 - data race in threadpool (Reported by Badalian
 * ASTERISK-26053 - res_pjsip_outbound_publish: Crash when shutting
      down (Reported by Joshua Colp)
 * ASTERISK-26049 - res_pjsip: Crash when our own request timer
      fires (Reported by Joshua Colp)
 * ASTERISK-25669 - [patch]CURL incorrect trim for non ASCII
      characters (Reported by Jesper)
 * ASTERISK-26029 - parking: ast_parking_park_call should return
      parking_space instead of parking_exten (Reported by Diederik de
 * 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 - [patch] problem with ringinuse=no, queue
      members receive sometimes two calls (Reported by nik600)
 * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir
      only works if you manually add secret.conf yourself (Reported by
      Jonathan R. Rose)
 * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions
      are case sensitive to QueueName (Reported by Javier Acosta)
 * ASTERISK-25951 - res_agi:  run_agi eats frames it shouldn't
      (Reported by George Joseph)
 * ASTERISK-25950 - [patch]SIP channel does not send PeerStatus
      events for autocreated peers (Reported by Kirill Katsnelson)
 * ASTERISK-25927 - Removed option "registertrying" is still
      documented in sip.conf.sample (Reported by Etienne Lessard)
 * ASTERISK-25947 - Protocol transfers to stasis applications are
      missing the StasisStart with the replace_channel object.
      (Reported by Richard Mudgett)
 * ASTERISK-24649 - Pushing of channel into bridge fails; Stasis
      fails to get app name (Reported by John Bigelow)
 * ASTERISK-24782 - StasisEnd event not present for channel that
      was swapped out for another after completing attended transfer
      (Reported by John Bigelow)
 * ASTERISK-25942 - res_pjsip_caller_id: Transfer results in mixed
      ConnectedLine information (Reported by George Joseph)
 * ASTERISK-25928 - res_pjsip: URI validation done outside of PJSIP
      thread (Reported by Joshua Colp)
 * ASTERISK-25929 - res_pjsip_registrar: AOR_CONTACT_ADDED events
      not raised (Reported by Joshua Colp)
 * ASTERISK-25934 - chan_sip should not require sipregs or
      updateable sippeers table unless rt (Reported by Jaco Kroon)
 * ASTERISK-25888 - Frequent segfaults in function can_ring_entry()
      of app_queue.c (Reported by Sébastien Couture)
 * ASTERISK-25796 - res_pjsip: DOS/Crash when TCP/TLS sockets
      exceed pjproject PJ_IOQUEUE_MAX_HANDLES (Reported by George
 * 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 - [patch] webrtc video broken due to missing
      marker bits in RTP streams (Reported by Jacek Konieczny)
 * ASTERISK-25881 - pbx: Add support for autohints (Reported by
      Joshua Colp)
 * ASTERISK-25854 - No audio after HOLD/RESUME - incorrect
      a=recvonly in SDP from Asterisk (Reported by Robert McGilvray)
 * ASTERISK-25868 - Sorcery "append to category" should allow
      filters (Reported by Nick Repin)
 * ASTERISK-25873 - res_pjsip: Bundled pjproject: compile error,
      cannot find -lasteriskpj (Reported by Hans van Eijsden)
 * ASTERISK-25882 - ARI: Crash can occur due to race condition when
      attempting to operate on a hung up channel (Part 2) (Reported by
      Richard Mudgett)
 * ASTERISK-25867 - [patch] Video delay on app_echo (Reported by
      Jacek Konieczny)
 * ASTERISK-24605 - res_parking option parkeddynamic does not work
      with the core Features 'parkcall' (DTMF initiated parking)
      (Reported by Philip Correia)
 * ASTERISK-24596 - Unclear how to use Park application with
      res_parking 'parkeddynamic' enabled. Documentation? (Reported by
      Philip Correia)
 * ASTERISK-25825 - Crashes during shutdown when running CLI
      commands (Reported by Mark Michelson)
 * ASTERISK-24543 - Asterisk 13 responds to SIP Invite with all
      possible codecs configured for peer as opposed to intersection
      of configured codecs and offered codecs (Reported by Taylor
 * ASTERISK-25407 - Asterisk fails to log to multiple syslog
      destinations (Reported by Elazar Broad)
 * ASTERISK-25510 - [patch]Log to syslog failing (Reported by
      Michael Newton)
 * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
      data corruption (Reported by Gianluca Merlo)
 * ASTERISK-25849 - chan_pjsip: transfers with direct media
      sometimes drops audio (Reported by Kevin Harwell)
 * ASTERISK-25814 - Segfault at f ip in res_pjsip_refer.so
      (Reported by Sergio Medina Toledo)
 * ASTERISK-25023 - Deadlock in chan_sip in
      update_provisional_keepalive (Reported by Arnd Schmitter)
 * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
      channel (Reported by Filip Frank)
 * ASTERISK-25829 - res_pjsip: PJSIP does not accept spaces when
      separating multiple AORs (Reported by Mateusz Kowalski)
 * ASTERISK-25771 - ARI:Crash - Attended transfers of channels into
      Stasis application. (Reported by Javier Riveros )
 * ASTERISK-25830 - Revision 2451d4e breaks NAT (Reported by Sean
 * 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 - [patch] Calculate talktime when is first call
      answered (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-25727 - RPM build requires OPTIONAL_API cflag due to
      PJSIP requirement (Reported by Gergely Dömsödi)
 * ASTERISK-25337 - Crash on PJSIP_HEADER Add P-Asserted-Identity
      when calling from Gosub (Reported by Jacques Peacock)
 * ASTERISK-25738 - res_pjsip_pubsub: Crash while executing
      OutboundSubscriptionDetail ami action (Reported by Kevin
 * ASTERISK-25721 - [patch] res_phoneprov: memory leak and
      heap-use-after-free (Reported by Badalian Vyacheslav)
 * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
      returns garbage (Reported by Etienne Lessard)
 * ASTERISK-25751 - res_pjsip: Support
      pjsip_dlg_create_uas_and_inc_lock (Reported by Joshua Colp)
 * ASTERISK-25606 - Core dump when using transports in sorcery
      (Reported by Martin 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 - [patch]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 - [patch]chan_sip: File descriptor leak with
      non-default timert1 (Reported by Alexander Traud)
 * ASTERISK-25702 - PjSip realtime DB and Cache Errors since
      upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported by
      Nic Colledge)
 * ASTERISK-25730 - build:  make uninstall after make distclean
      tries to remove root (Reported by George Joseph)
 * ASTERISK-25725 - core: Incorrect XML documentation may result in
      weird behavior (Reported by Joshua Colp)
 * ASTERISK-25722 - ASAN & testsute: stack-buffer-overflow in
      sip_sipredirect (Reported by Badalian Vyacheslav)
 * ASTERISK-25709 - ARI: Crash can occur due to race condition when
      attempting to operate on a hung up channel (Reported by Mark
 * 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 - [patch]Issue a TCP connection(kernel) and
      thread of asterisk is not released (Reported by Hiroaki Komatsu)
 * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
      Corey Farrell)
 * ASTERISK-25619 - res_chan_stats not sending the correct
      information to StatsD (Reported by Tyler Cambron)
 * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
      answer waiting time is more than ~7sec (Reported by Aleksei
 * ASTERISK-25609 - [patch]Asterisk may crash when calling
      ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
 * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
      (Reported by Alexander Traud)
 * ASTERISK-25616 - Warning with a Codec Module which supports PLC
      with FEC (Reported by Alexander Traud)
 * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
      Dudás József)
 * ASTERISK-25608 - res_pjsip/contacts/statsd:  Lifecycle events
      aren't consistent (Reported by George Joseph)
 * ASTERISK-25584 - [patch] format-attribute module: VP8 missing
      (Reported by Alexander Traud)
 * ASTERISK-25583 - [patch] format-attribute module: RFC 7587 (Opus
      Codec) (Reported by Alexander Traud)
 * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
      that module installed (Reported by Ben Langfeld)
 * ASTERISK-25595 - Unescaped : in messge sent to statsd (Reported
      by Niklas Larsson)
 * ASTERISK-25598 - res_pjsip:  Contact status messages are
      printing a hash instead of the uri (Reported by George Joseph)
 * ASTERISK-25600 - bridging: Inconsistency in BRIDGEPEER (Reported
      by Jonathan Rose)
 * ASTERISK-25476 - chan_sip loses registrations after a while
      (Reported by Michael Keuter)
 * ASTERISK-25593 - fastagi: record file closed after sending
      result (Reported by Kevin Harwell)
 * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
      it's assumed to (Reported by Walter Doekes)
 * ASTERISK-25590 - CLI Usage info for 'pjsip send notify'
      references incorrect config (Reported by Corey Farrell)
 * ASTERISK-25165 - Testsuite - Sorcery memory cache leaks
      (Reported by Corey Farrell)
 * ASTERISK-25575 - res_pjsip: Dynamic outbound registrations
      created via ARI are not loaded into memory on Asterisk
      start/restart (Reported by Matt Jordan)
 * ASTERISK-25545 - [patch] translation module gets cached not
      joint format (Reported by Alexander Traud)
 * ASTERISK-25573 - [patch] H.264 format attribute module: resets
      whole SDP (Reported by Alexander Traud)
 * ASTERISK-24958 - Forwarding loop detection inhibits certain
      desirable scenarios (Reported by Mark Michelson)
 * ASTERISK-25561 - app_queue.c line 6503 (try_calling): mutex
      'qe->chan' freed more times than we've locked! (Reported by Alec
 * ASTERISK-25565 - DNS: System resolver only returns 1 record per
      result (Reported by George Joseph)
 * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
      Joshua Colp)
 * ASTERISK-25160 - [patch] Opus Codec: SIP/SDP line fmtp missing
      when called internally (Reported by Alexander Traud)
 * ASTERISK-25535 - [patch] format creation on module load instead
      of cache (Reported by Alexander Traud)
 * ASTERISK-25449 - main/sched: Regression introduced by
      5c713fdf18f causes erroneous duplicate RTCP messages; other
      potential scheduling issues in chan_sip/chan_skinny (Reported by
      Matt Jordan)
 * ASTERISK-25546 - threadpool: Race condition between idle timeout
      and activation (Reported by Joshua Colp)
 * ASTERISK-25537 - [patch] format-attribute module: RFC or
      internal defaults? (Reported by Alexander Traud)
 * ASTERISK-25533 - [patch] buffer for ast_format_cap_get_names
      only 64 bytes (Reported by Alexander Traud)
 * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
      the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
 * 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 - [patch] Update the maximum packetization values
      in frame.c (Reported by dea)
 * ASTERISK-25484 - [patch] autoframing=yes has no effect (Reported
      by Alexander Traud)
 * ASTERISK-25308 - ari: Websocket leak (Reported by Joshua Colp)
 * ASTERISK-25461 - Nested dialplan #includes don't work as
      expected. (Reported by Richard Mudgett)
 * ASTERISK-25455 - Deadlock of PJSIP realtime over
      res_config_pgsql  (Reported by mdu113)
 * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
      (Reported by Olle Johansson)
 * ASTERISK-25108 - configure check for older unbound library
      (Reported by John Bigelow)
 * ASTERISK-25435 - Asterisk periodically hangs. UDP Recv-Q greatly
      exceeds zero. (Reported by Dmitriy Serov)
 * ASTERISK-25451 - Broken video - erased rtp marker bit (Reported
      by Stefan Engström)
 * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
      exist in AstDB (Reported by Andrew Nagy)
 * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
      parsing (Reported by ffs)
 * ASTERISK-25404 - segfault/crash in chan_pjsip_hangup ... at
      chan_pjsip.c (Reported by Chet Stevens)
 * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
      (Reported by Bojan 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 - [patch] Transcoding while different in Frame
      size = Frames lost (Reported by Alexander Traud)
 * ASTERISK-25325 - ARI PUT reload chan_sip HTTP response 404
      (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-25390 - default_from_user can crash with certain
      configuration backends (Reported by Mark Michelson)
 * ASTERISK-25387 - res_pjsip_nat: Malformed REGISTER request
      causes NAT'd Contact header to not be rewritten (Reported by
      Matt Jordan)
 * ASTERISK-25227 - No audio at in-band announcements in ooh323
      channel (Reported by Alexandr Dranchuk)
 * ASTERISK-25295 - res_pjsip crash - pjsip_uri_get_uri at
      /usr/include/pjsip/sip_uri.h (Reported by Dmitriy Serov)
 * ASTERISK-25381 - res_pjsip: AoRs deleted via ARI (or other
      mechanism) do not destroy their related contacts (Reported by
      Matt Jordan)
 * ASTERISK-25369 - res_parking: ParkAndAnnounce - Inheritable
      variables aren't applied to the announcer channel (Reported by
      Jonathan Rose)
 * ASTERISK-25367 - pbx: Long pattern match hints may cause "core
      show hints" to crash (Reported by Joshua Colp)
 * ASTERISK-25365 - Persistent subscriptions have extra
      Content-Length/corrupted messages (Reported by Mark Michelson)
 * ASTERISK-25356 - res_pjsip_sdp_rtp: Multiple keepalive scheduled
      items may exist (Reported by Joshua Colp)
 * ASTERISK-25355 - sched: ast_sched_del may return prematurely due
      to spurious wakeup (Reported by Joshua Colp)
 * ASTERISK-25318 -
      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 - [patch] iLBC 20 advertised (Reported by
      Alexander Traud)
 * ASTERISK-25304 - res_pjsip: XML sanitization may write past
      buffer (Reported by Joshua Colp)
 * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
      Firefox 39 - add ECDH support and fallback to prime256v1
      (Reported by Stefan Engström)
 * ASTERISK-24988 - func_talkdetect: Test is bouncing sporadically
      (Reported by Joshua Colp)
 * ASTERISK-25181 - ARI: Channels added to Stasis application
      during WebSocket creation don't receive a StasisStart event
      (Reported by Matt Jordan)
 * ASTERISK-25296 - RTP performance issue with several channel
      drivers. (Reported by Richard Mudgett)
 * ASTERISK-25297 - Crashes running
      channels/pjsip/resolver/srv/failover/in_dialog testsuite tests
      (Reported by Richard Mudgett)
 * ASTERISK-25292 - Testuite:
      tests/apps/bridge/bridge_wait/bridge_wait_e_options fails
      (Reported by Kevin Harwell)
 * ASTERISK-25271 - Parking & blind transfer: Transferer channel
      not hung up if no MOH (Reported by Kevin Harwell)
 * ASTERISK-25250 - chan_sip - Despite the channel being answered,
      caller on a call established via Local channel continues to hear
      ringback (Reported by Etienne Lessard)
 * ASTERISK-25253 - confbridge volume options and other volume
      controls such as func_volume don't work (Reported by Dmitriy
 * ASTERISK-25247 - choppy audio when spying on a g722 channel,
      chan_sip or chan_pjsip (Reported by hristo)
 * ASTERISK-25263 - [patch]cdr_adaptive_odbc: CDR insert failure
      due to reversed if logic (Reported by Elazar Broad)
 * ASTERISK-24867 - Docs for 'e' option in ResetCDR say to use
      CDR_PROP instead, CDR_PROP docs are unclear (Reported by Rusty
 * ASTERISK-24853 - Documentation claims chan_sip outbound
      registrations support WS or WSS as valid transports (not true)
      (Reported by PSDK)
 * ASTERISK-25242 - PJSIP: No audio when Asterisk inside NAT and
      endpoints outside NAT - implement functionality similar to
      chan_sip 'rtpkeepalive'? (Reported by Mark Michelson)
 * ASTERISK-25258 - chan_pjsip: Incorrect format switch on received
      RTP packet (Reported by Joshua Colp)
 * ASTERISK-25257 - [patch]channels/sig_pri.h -> sig_pri_span ->
      force_restart_unavailable_chans in wrong scope (Reported by
      Patric Marschall)
 * ASTERISK-24934 - [patch]Asterisk manager output does not escape
      control characters (Reported by warren smith)
 * ASTERISK-25255 - Missing AMI VarSet events when setting to an
      empty string. (Reported by Richard Mudgett)
 * ASTERISK-25254 - Crash if dialplan sets ATTENDEDTRANSFER to an
      empty string before Park. (Reported by Richard Mudgett)
 * ASTERISK-25183 - PJSIP: Crash on NULL channel in
      chan_pjsip_incoming_response despite previous checks for NULL
      channel (Reported by Matt Jordan)
 * ASTERISK-25201 - Crash in PJSIP distributor on already free'd
      threadpool (Reported by Matt Jordan)
 * ASTERISK-25240 - bridge_native_rtp: Direct media wrongfully
      started when completing attended transfer (Reported by Joshua
 * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
      (Reported by Rusty Newton)
 * ASTERISK-25146 - DNS: Create system level resolver (Reported by
      Joshua Colp)
 * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
      BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
      (Reported by Dmitry Burilov)
 * ASTERISK-24550 - res_rtp_asterisk: Crash in
      ast_rtp_on_ice_complete during DTLS handshake (Reported by
      Osaulenko Alexander)
 * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
      Badalian Vyacheslav)
 * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
      by Stefan Engström)
 * ASTERISK-25127 - DTLS crashes following "Unable to cancel
      schedule ID" in dtls_srtp_check_pending (Reported by Dade
 * 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 - [patch]Closing of fd -1 in chan_mgcp.c
      (Reported by Walter Doekes)
 * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
      in rtp_engine.c (Reported by Walter Doekes)
 * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
      (Reported by Walter Doekes)
 * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
      Bad file descriptor" (Reported by Barry Chern)
 * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
      13.4 (Reported by cervajs)
 * ASTERISK-25196 - res_pjsip_nat: rewrite_contact should not be
      applied to Contact header when Record-Route headers are present
      (Reported by Mark Michelson)
 * ASTERISK-24907 - res_pjsip_outbound_registration: crash during
      unload if registration attempts are still occuring (Reported by
      Kevin Harwell)
 * ASTERISK-25204 - res_pjsip_refer: Duplicated Referred-By or
      Replaces headers on outbound INVITEs. (Reported by Mark
 * ASTERISK-25189 - AMI: Add Linkedid header to standard channel
      snapshot information. (Reported by Richard Mudgett)
 * ASTERISK-25171 - Early completion of feature code attended
      transfer results in intermittent one-way audio, "ghost ringing"
      and robotic sound. (Reported by Rusty Newton)
 * ASTERISK-25172 - Crash in channels/sip/sip blind
      transfer/caller_refer_only test in
      ast_format_cap_append_from_cap during ast_request (Reported by
      Matt Jordan)
 * ASTERISK-25180 - res_pjsip_mwi: Unsolicited MWI requires reload
      (Reported by Joshua Colp)
 * ASTERISK-25182 - [patch] on CLI sip reload, new codecs get
      appended only (Reported by Alexander Traud)
 * ASTERISK-25163 - Deadlock in chan_sip between reload of sip peer
      container and MWI Stasis callback (Reported by Dmitriy Serov)
 * ASTERISK-25091 - Asterisk REST API - bridge.addChannel crash
      asterisk when calling channel hangup while adding to bridge
      (Reported by Ilya Trikoz)
 * ASTERISK-24900 - Manager event ParkedCallSwap is not documented
      (Reported by Rusty Newton)
 * ASTERISK-25162 - func_pjsip_aor: Leak of contact in iterator
      (Reported by Corey Farrell)
 * ASTERISK-25158 - res_pjsip: Add option to use AAL2 packing when
      negotiating g.726 (Reported by Kevin Harwell)
 * ASTERISK-24344 - CDR_PROP(disable) disables CDR only for first
      dialed party (Reported by Janusz Karolak)
 * ASTERISK-24443 - CDR fields (dst, dcontext) empty in transfer
      call started from Macro (Reported by Arveno Santoro)
 * ASTERISK-25154 - [patch]fromtag may need to be updated after
      successful call dialog match (Reported by Damian Ivereigh)
 * ASTERISK-25156 - chan_pjsip’s CHAN_START cel event lacks the
      correct context and exten (Reported by cloos)
 * ASTERISK-25157 - bridging: Performing a blonde transfer does not
      result in connected line updates (Reported by Joshua Colp)
 * ASTERISK-25087 - Asterisk segfault when using Directory
      application with alias option and specific mailbox configuration
      (Reported by Chet Stevens)
 * ASTERISK-25115 - Crash related to func
      sip_resolve_invoke_user_callback of res_pjsip/pjsip_resolver.c
      (Reported by John Bigelow)
 * ASTERISK-25096 - [patch]Segfault when registering over
      websockets with PJSIP (in ast_sockaddr_isnull at
      /include/asterisk/netsock2.h) (Reported by Josh Kitchens)
 * ASTERISK-24963 - ASAN: heap-use-after-free with PJSIP and WSS
      (Reported by Badalian Vyacheslav)
 * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
      but asterisk doesn't detect it. (Reported by ibercom)
 * ASTERISK-25094 - PBX core: Investigate thread safety issues
      (Reported by Corey Farrell)
 * ASTERISK-25113 - install_prereq in Debian 8 without "standard
      system utilities" (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-25148 - res_pjsip NULL channel audit (Reported by Mark
 * ASTERISK-25131 - chan_pjsip: In-dialog authentication not
      handled. (Reported by Richard Mudgett)
 * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
      | adpcm | ipc10} (Reported by Badalian Vyacheslav)
 * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
      that end with ::80 (Reported by Mark Petersen)
 * ASTERISK-25122 - Large SIP packet received via pjsip over
      websocket crashes Asterisk  (Reported by Ivan Poddubny)
 * ASTERISK-25121 - Stasis: Fix unsafe use of stasis_unsubscribe in
      modules. (Reported by Corey Farrell)
 * ASTERISK-25120 - Astobj2: Weakproxy subscriptions should be run
      in reverse order. (Reported by Corey Farrell)
 * ASTERISK-25105 - res_pjsip:  Possible incompatibility between
      qualify_timeout and pjproject-2.4 (Reported by George Joseph)
 * ASTERISK-25117 - res_mwi_external_ami: Fix manager action
      registrations. (Reported by Corey Farrell)
 * ASTERISK-25112 - Logger: Configuration settings are not reset to
      default during reload. (Reported by Corey Farrell)
 * ASTERISK-24983 - IAX deadlock between hangup and scheduled
      actions (ex. largrq) (Reported by Y Ateya)
 * ASTERISK-24944 - main/audiohook.c change prevents G722 call
      recording (Reported by Ronald Raikes)
 * ASTERISK-25110 - res_resolver_unbound.c compilation failure:
      SIGURG is undeclared in func unbound_resolver_stop (Reported by
      John Bigelow)
 * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
      or more digits (Reported by Makoto Dei)
 * ASTERISK-25086 - [patch]PJSIP crashes if endpoint missing in
      Dial() (Reported by snuffy)
 * ASTERISK-25089 - res_pjsip_config_wizard: Variable specified in
      templates aren't being processed correctly (Reported by George
 * 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 - [patch]Potential crash after unload of
      func_periodic_hook or test_message (Reported by Corey Farrell)
 * ASTERISK-25082 - Asterisk deletes message after doing a playback
      of an INBOX message using ast_vm_play when the Old folder is
      full for that mailbox. (Reported by Jonathan Rose)
 * ASTERISK-21893 - Segfault after call hangup, in
      ast_channel_hangupcause_set, at channel_internal_api.c (Reported
      by Aleksandr Gordeev)
 * ASTERISK-25042 - asterisk.conf options override command-line
      options. (Reported by Corey Farrell)
 * ASTERISK-25074 - Regression: Recent clang-related change broke
      cross compiling of Asterisk (Reported by Sebastian Kemper)
 * ASTERISK-24442 - Outgoing call files don't work properly when
      set in the future (Reported by tootai)
 * ASTERISK-18252 - queue_log mysql time column data format
      (Reported by Gareth Blades)
 * ASTERISK-25041 - [patch]Broken column type checking in
      res_config_mysql addon (Reported by Alexandre Fournier)
 * ASTERISK-25057 - res_pjsip_pubsub: Crash in send_notify due to
      invalid root pointer in sub_tree (Reported by Matt Jordan)
 * ASTERISK-24938 - ARI Snoop Channel results in excessive
      escalating CPU usage (Reported by George Ladoff)
 * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
      ignore ISDN RESTART requests. (Reported by Richard Mudgett)
 * ASTERISK-25003 - Asterisk crashes on attended transfer (using
      feature) (Reported by Artem Volodin)
 * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
      contain waiting time (Reported by Etienne Lessard)
 * ASTERISK-25027 - Build System: Many ARI modules are missing
      dependencies. (Reported by Corey Farrell)
 * ASTERISK-25061 - pbx_config: Register manager actions with
      module version of macro. (Reported by Corey Farrell)
 * ASTERISK-24967 - Problem support schema for pgsql on CEL
      (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-25025 - Periodic crashes (in
      ast_channel_snapshot_create at stasis_channels.c) with Certified
      Asterisk 13. (Reported by Chet Stevens)
 * ASTERISK-25053 - Unit test category /main/presence missing
      trailing slash. (Reported by Corey Farrell)
 * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
      not respected, failover between DSNs doesn't work (Reported by
 * 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 - [patch] Using force black background leads to
      colours not being reset (Reported by dant)
 * ASTERISK-25048 - Astobj2: Initialization order wrong when both
      refdebug and AO2_DEBUG are both enabled. (Reported by Corey
 * 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 - [patch]CLI command cdr show pgsql status
      (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-20524 - AMI improperly handles lines of exactly 1025
      characters (Reported by David M. Lee)
 * ASTERISK-24936 - New Feature: AO2 weakproxy objects (Reported by
      Corey Farrell)
 * ASTERISK-24954 - Git migration: Asterisk version numbers are
      incompatible with the Test Suite (Reported by Matt Jordan)
 * ASTERISK-17608 - func_aes.so cannot be loaded if res_crypto /
      openssl not compiled (Reported by Warren Selby)
 * ASTERISK-24928 - [patch]t38_udptl_maxdatagram in pjsip.conf not
      honored (Reported by Juergen Spies)
 * ASTERISK-24835 - Early Media Not working with Chan SIP and
      Asterisk 13 (Reported by Andrew Nagy)
 * ASTERISK-21777 - Asterisk tries to transcode video instead of
      audio (Reported by Nick Ruggles)
 * ASTERISK-24380 - core: Native formats are set to h264 with
      certain audio/video codec configuration, resulting in path
      translation WARNINGs (Reported by Matt Jordan)
 * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
      into account (Reported by Frederic Van Espen)
 * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
      short (Reported by Y Ateya)
 * ASTERISK-24935 - res_pjsip_phoneprov_provider: Fix leaked
      OBJ_MULTIPLE iterator. (Reported by Corey Farrell)
 * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
      (Reported by Vadim)
 * ASTERISK-24933 - T38 fails negotiation (Reported by Jonathan
 * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
      byte prefix bug (Reported by Matt Jordan)
 * ASTERISK-21211 - chan_iax2 - unprotected access of
      iaxs[peer->callno] potentially results in segfault (Reported by
      Jaco Kroon)
 * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
      (Reported by Christoph Timm)
 * ASTERISK-24910 - "timer=no" and "timer=required" settings in
      pjsip.conf fail (Reported by Ray Crumrine)
 * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
      (Reported by Jeffrey C. Ollie)
 * ASTERISK-24914 - Division by zero in file.c when playback of
      voicemail with video as h264 (Reported by Marcello Ceschia)
 * ASTERISK-24899 - Parking fall-through behavior different in 13
      (Reported by Malcolm Davenport)
 * ASTERISK-24937 - [patch]res_pjsip_messaging: Messages may be
      sent out of order (Reported by Mark Michelson)
 * ASTERISK-24920 - Asterisk handles duplicate SIP requests as if
      they were each a new request (Reported by Mark Michelson)
 * ASTERISK-24781 - PJSIP: Unnecessary 180 Ringing messages sent
      with undesireabe consequences. (Reported by Richard Mudgett)
 * ASTERISK-24857 - [patch] "timing test", pjsip incoming/outgoing
      calls, voicemail prompts and recordings all fail when using the
      kqueue timer source on FreeBSD 10.x (Reported by Justin T.
 * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
      detection in ast_malloc (Reported by Timo Teräs)
 * ASTERISK-24142 - CCSS: crash during shutdown due to device
      lookup in destroyed container (Reported by David Brillert)
 * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
      core restart now (Reported by Peter Katzmann)
 * ASTERISK-24805 - [patch] - ASAN: Race condition
      (heap-use-after-free) on asterisk closing (Reported by Badalian
 * ASTERISK-24881 - ast_register_atexit should only be used when
      absolutely needed (Reported by Corey Farrell)
 * ASTERISK-24731 - res_pjsip_session cannot be unloaded (Reported
      by Corey Farrell)
 * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
      (Reported by Kevin Harwell)
 * ASTERISK-14233 - [patch] Buddies are always auto-registered when
      processing the roster (Reported by Simon Arlott)
 * ASTERISK-24780 - [patch] - Buddies are always auto-registered
      when processing the roster (Reported by Simon Arlott)
 * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
      under OpenBSD (Reported by snuffy)
 * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
 * ASTERISK-21765 - [patch] - FILE function's length argument
      counts from beginning of file rather than the offset (Reported
      by John Zhong)
 * ASTERISK-24817 - init_logger_chain: unreachable code block
      (Reported by Corey Farrell)
 * ASTERISK-24882 - chan_sip: Improve usage of REF_DEBUG (Reported
      by Corey Farrell)
 * ASTERISK-24876 - Investigate reference leaks from
      tests/channels/local/local_optimize_away (Reported by Corey
 * ASTERISK-24840 - res_pjsip: conflicting endpoint identifiers
      (Reported by Kevin Harwell)
 * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
      Atis Lezdins)
 * ASTERISK-18708 - func_curl hangs channel under load (Reported by
      Dave Cabot)
 * ASTERISK-21038 - Bad command completion of "core set debug
      channel" (Reported by Richard Kenner)
 * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
      by Frank DiGennaro)
 * ASTERISK-24872 - [patch] AMI PJSIPShowEndpoint closes AMI
      connection on error (Reported by Dmitriy Serov)
 * ASTERISK-23666 - CLONE - nested functions aren't portable
      (Reported by Diederik de Groot)
 * ASTERISK-20399 - Compilation on some systems requires the
      -fnested-functions flag (Reported by David M. Lee)
 * ASTERISK-20850 - [patch]Nested functions aren't portable.
      Adapting RAII_VAR to use clang/llvm blocks to get the
      same/similar functionality. (Reported by Diederik de Groot)
 * ASTERISK-24807 - Missing mandatory field Max-Forwards (Reported
      by Anatoli)
 * ASTERISK-24808 - res_config_odbc: Improper escaping of
      backslashes occurs with MySQL (Reported by Javier Acosta)
 * ASTERISK-23390 - NewExten Event with application AGI shows up
      before and after AGI runs (Reported by Benjamin Keith Ford)
 * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
      voicemail stored in LDAP (Reported by Graham Barnett)
 * ASTERISK-24739 - [patch] - Out of files -- call fails --
      numerous files with inodes from under /usr/share/zoneinfo,
      mostly posixrules (Reported by Ed Hynan)
 * ASTERISK-24755 - Asterisk sends unexpected early BYE to
      transferrer during attended transfer when using a Stasis bridge
      (Reported by John Bigelow)
 * ASTERISK-24830 - res_rtp_asterisk.c checks USE_PJPROJECT not
      HAVE_PJPROJECT (Reported by Stefan Engström)
 * ASTERISK-24825 - Caller ID not recognized using
      Centrex/Distinctive dialing (Reported by Richard Mudgett)
 * ASTERISK-17588 - Caller ID on TDM410P *UK* PSTN (Reported by
      Daniel Flounders)
 * ASTERISK-24838 - chan_sip: Locking inversion occurs when
      building a peer causes a peer poke during request handling
      (Reported by Richard Mudgett)
 * ASTERISK-24751 - Integer values in json payload to ARI cause
      asterisk to crash (Reported by jeffrey putnam)
 * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
 * ASTERISK-18105 - most of asterisk modules are unbuildable in
      cygwin environment (Reported by feyfre)
 * ASTERISK-21845 - maxcalls exceeded, Asterisk sends out 480 and
      also BYE (Reported by Tony Ching)
 * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
      error response and BYE are sent to the caller (Reported by
      Makoto Dei)
 * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
      SRTP for audio, but they responded without it' is ambiguous and
      wrong in some cases (Reported by Rusty Newton)
 * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
      fail (Reported by Terry Wilson)
 * ASTERISK-20233 - SRTP not working with some devices (Eg
      Grandstream gxv3175) - Message "Can't provide secure audio
      requested in SDP offer" (Reported by tootai)
 * ASTERISK-22748 - SRTP Crypto Offer With Lifetime Not Accepted
      (Reported by Alejandro Mejia)
 * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
      thread ID being passed to pthread_kill (Reported by JoshE)
 * ASTERISK-24812 - ARI: Creating channels through /channels
      resource always uses SLIN, which results in unneeded transcoding
      (Reported by Matt Jordan)
 * ASTERISK-24797 - bridge_softmix: G.729 codec license held
      (Reported by Kevin Harwell)
 * ASTERISK-24677 - ARI GET variable on channel provides unhelpful
      response on non-existent variable (Reported by Joshua Colp)
 * ASTERISK-24785 - 'Expires' header missing from 200 OK on
      REGISTER (Reported by Ross Beer)
 * ASTERISK-24499 - Need more explicit debug when PJSIP dialstring
      is invalid (Reported by Rusty Newton)
 * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
      (Reported by Ashley Sanders)
 * ASTERISK-24796 - Codecs and bucket schema's prevent module
      unload (Reported by Corey Farrell)
 * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
      OSX with 64 bit integers (Reported by Corey Farrell)
 * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
      for playing back messages stored in IMAP - play_message: No
      origtime (Reported by Graham Barnett)
 * ASTERISK-22670 - Asterisk crashes when processing ISDN AoC
      Events (Reported by klaus3000)
 * ASTERISK-24689 - Segfault on hangup after outgoing PRI-Euroisdn
      call (Reported by Marcel Manz)
 * ASTERISK-24740 - [patch]Segmentation fault on aoc-e event
      (Reported by Panos Gkikakis)
 * ASTERISK-24799 - [patch] make fails with undefined reference to
      SSLv3_client_method (Reported by Alexander Traud)
 * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
      (Reported by Corey Farrell)
 * ASTERISK-24700 - CRASH: NULL channel is being passed to
      ast_bridge_transfer_attended() (Reported by Zane Conkle)
 * ASTERISK-24791 - Crash in ast_rtcp_write_report (Reported by
 * 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 - [patch] Fix ast_odbc_find_table function in
      res_odbc (Reported by ibercom)
 * ASTERISK-24769 - res_pjsip_sdp_rtp: Local ICE candidates leaked
      (Reported by Matt Jordan)
 * ASTERISK-24748 - res_pjsip: If wizards explicitly configured in
      sorcery.conf false ERROR messages may occur (Reported by Joshua
 * 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 - [patch] chan_sip realtime uses case sensitive
      column comparison for 'defaultuser' (Reported by
 * ASTERISK-24709 - [patch] 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
      chan_sip.so) (Reported by Stefan Engström)
 * ASTERISK-24640 - Registration pending stays forever after sip
      reload (Reported by Max Man)
 * ASTERISK-24682 - app_dial: Multiple DialEnd events emitted when
      MACRO_RESULT or GOSUB_RESULT are an unexpected value (Reported
      by Matt Jordan)
 * ASTERISK-24560 - Creating a named ARI bridge twice causes a
      crash (Reported by Kinsey Moore)
 * ASTERISK-24600 - Stuck IAX channels, Asterisk stops responding
      to most traffic, potential deadlock (Reported by Jeff Collell)
 * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
      32-bit packages on 64-bit hosts (Reported by Ben Klang)
 * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
      voicemail is not deleted after review, hangup (Reported by LEI
 * ASTERISK-24615 - When Multiple Transports Exist in pjsip.conf,
      Incorrect External Addresses is Used in SIP Packets When
      Responding to INVITE (Reported by David Justl)
 * ASTERISK-24624 - Transfer to invalid extension results in hung
      channel. (Reported by Zane Conkle)
 * ASTERISK-24663 - [patch] Unnamed semaphore autoconf check fails
      on cross compilation (Reported by abelbeck)
 * ASTERISK-24655 - res_pjsip_outbound_publish: Hang on shutdown
      while attempting to publish (Reported by Kevin Harwell)
 * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
      in the CFlags returned (Reported by Diederik de Groot)
 * ASTERISK-23850 - Park Application does not respect Return
      Context Priority (Reported by Andrew Nagy)
 * ASTERISK-24665 - Configure check required for
      pjsip_get_dest_info() (Reported by Mark Michelson)
 * ASTERISK-24049 - Asterisk Manager Interface: A number of list
      type responses aren't using astman_send_listack (Reported by
      Jonathan Rose)
 * ASTERISK-20744 - [patch] Security event logging does not work
      over syslog (Reported by Michael Keuter)
 * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
      (Reported by Kristian Høgh)
 * ASTERISK-24474 - sip_to_pjsip.py lacks documentation and does
      not function (Reported by John Kiniston)
 * ASTERISK-24637 - Channel re-enters Stasis() when it should not
      (Reported by John Bigelow)
 * ASTERISK-24591 - Stasis() side of an ARI originated channel
      cannot be Redirected (Reported by Kinsey Moore)
 * ASTERISK-24376 - res_pjsip_refer: REFER request for remote
      session attempts to direct channel to external_replaces
      extension instead of context, without providing for the
      Referred-To SIP URI (Reported by Matt Jordan)
 * ASTERISK-24513 - Local channel apparently leaked in off-nominal
      DTMF attended transfer (Reported by Mark Michelson)
 * ASTERISK-24367 - PJSIP: allow all results in failure to send
      INVITE (Reported by Scott Griepentrog)
 * ASTERISK-24267 - Queue variables associated with
      setinterfacevar, setqueueentryvar, setqueuevar are not passed to
      local channel (Reported by Mitch Claborn)
 * ASTERISK-24641 - Deadlock in Trunk (Reported by Malcolm
 * ASTERISK-23841 - DTMF atxfer doesn't set CallerID for the recall
      calls to the transferrer. (Reported by Richard Mudgett)
 * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
      destination when 'sendrpid=yes' (in proxy environment) (Reported
      by Karsten Wemheuer)
 * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
      on startup (Reported by Richard Kenner)
 * ASTERISK-24566 - Uninit buf in WS write (Reported by Badalian
 * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
      level - 'Remote address is null, most likely RTP has been
      stopped' (Reported by Rusty Newton)
 * ASTERISK-24459 - bridge_native_rtp: Native RTP bridging is
      chosen for RTP compatible channels when the DTMF mode is not
      compatible (Reported by Yaniv Simhi)
 * ASTERISK-24536 - AMI redirect with PJSIP fails to move extra
      channel (Reported by Niklas Larsson)
 * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
      casts char to unsigned int (Reported by Walter Doekes)
 * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
      enabled (Reported by Andreas Steinmetz)
 * ASTERISK-22455 - Asterisk 12 on Ubuntu Lucid deadlocks with
      DEBUG_THREADS+OPTIONAL_API enabled (Reported by David M. Lee)
 * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
      enabled (Reported by Richard Mudgett)
 * ASTERISK-24604 - res_rtp_asterisk: Crash during restart due to
      race condition in accessing codec in stored ast_frame and codec
      core (Reported by Matt Jordan)
 * ASTERISK-24563 - Direct Media calls within private network
      sometimes get one way audio (Reported by Kevin Harwell)
 * ASTERISK-24607 - res_pjsip_session: re-INVITE with declined
      media streams results in 488 (Reported by Matt Jordan)
 * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
      from JSSIP (Reported by Badalian Vyacheslav)
 * ASTERISK-24514 - res_pjsip_outbound_registration: stack overflow
      when using non-default sorcery wizard (Reported by Kevin
 * ASTERISK-24342 - PJSIP: Qualifying endpoints attempts to do them
      all at the same time. (Reported by Richard Mudgett)
 * ASTERISK-24556 - Asterisk 13 core dumps when calling from pjsip
      extension to another pjsip extension  (Reported by Abhay Gupta)
 * ASTERISK-24537 - Stasis: StasisStart/StasisEnd events are not
      reliably transmitted during transfers (Reported by Matt Jordan)
 * ASTERISK-24573 - [patch]Out of sync conversation recording when
      divided in multiple recordings (Reported by Nuno Borges)
 * ASTERISK-24572 - [patch]App_meetme is loaded without its
      defaults when the configuration file is missing (Reported by
      Nuno Borges)
 * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
      voicemail under high concurrency with an IMAP backend (Reported
      by David Duncan Ross Palmer)
 * ASTERISK-24274 - [patch]Codec Format Is Not Included in the SDP
      Media Attributes When SLIN48 Codec Is Used (Reported by Frankie
 * ASTERISK-24533 - 2 threads created per chan_sip entry (Reported
      by xrobau)
 * ASTERISK-24542 - [patch]Failure showing codecs via 'core show
      channeltype <tech>' (Reported by snuffy)
 * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
      allow blocked addresses through (Reported by Matt Jordan)
 * ASTERISK-24534 - [patch]Register DB() as escalating to prevent
      users from writing to astdb (Reported by Gareth Palmer)
 * ASTERISK-24531 - res_pjsip_acl: ACLs not applied on initial
      module load (Reported by Matt Jordan)
 * ASTERISK-24490 - Security Vulnerability: CONFBRIDGE function's
      record_command option allows arbitrary parameters to be passed
      to MixMonitor, allowing remote execution of commands (Reported
      by Matt Jordan)
 * ASTERISK-24528 - res_pjsip_refer: Sending INVITE with Replaces
      in-dialog with invalid target causes crash (Reported by Joshua
 * ASTERISK-24471 - Crash - assert_fail in libc in
      pjmedia_sdp_neg_negotiate from /usr/local/lib/libpjmedia.so.2
      (Reported by yaron nahum)
 * ASTERISK-24535 - stringfields: Fix regression from fix for
      unintentional memory retention and another issue exposed by the
      fix (Reported by Corey Farrell)
 * ASTERISK-24508 - pjsip - REFER request from SNOM is rejected
      with "400 bad request" - DEBUG shows "Received a REFER without a
      parseable Refer-To" (Reported by Beppo Mazzucato)
 * ASTERISK-15242 - transmit_refer leaks sip_refer structures
      (Reported by David Woolley)
 * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
      endmarked users when last marked user leaves (Reported by Matt
 * 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 - [patch] Voicemail with multi-recipients To:
      header fix (Reported by abelbeck)
 * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
      (Reported by Corey Farrell)
 * ASTERISK-24447 - Bridge DTMF hooks: Audio doesn't pass when
      waiting for more matching digits. (Reported by Richard Mudgett)
 * ASTERISK-24257 - agent must dial acceptdtmf twice to bridge to
      queue caller (Reported by Steve Pitts)
 * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
      extra calls to ast_module_unref (Reported by Corey Farrell)
 * ASTERISK-24491 - Memory leak in res_hep (Reported by Zane
 * ASTERISK-24307 - Unintentional memory retention in stringfields
      (Reported by Etienne Lessard)
 * ASTERISK-24438 - res_pjsip_multihomed.so blocks Asterisk reload
      when DNS settings invalid (Reported by Melissa Shepherd)
 * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
      unescapes semicolons ("\;" -> ";") turning them into comments
      (corruption) on rewrite of a config file (Reported by George
 * ASTERISK-24487 - configuration: sections should be loadable as
      template even when not marked (Reported by Scott Griepentrog)
 * ASTERISK-24482 - func_talkdetect: Fix stasis message leak in
      audiohook callback (Reported by Corey Farrell)
 * ASTERISK-24480 - res_http_websockets: Module reference decrease
      below zero (Reported by Corey Farrell)
 * ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream
      leaks (Reported by Corey Farrell)
 * ASTERISK-24411 - [patch] Status of outbound registration is not
      changed upon unregistering. (Reported by John Bigelow)
 * ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled
      (Reported by Corey Farrell)
 * ASTERISK-24466 - app_queue: fix a couple leaks to struct
      call_queue (Reported by Corey Farrell)
 * ASTERISK-24465 - audiohooks list leaks reference to formats
      (Reported by Corey Farrell)
 * ASTERISK-24462 - res_pjsip: Stale qualify statistics after
      disablementation (Reported by Kevin Harwell)
 * ASTERISK-24190 - IMAP voicemail causes segfault (Reported by
      Nick Adams)
 * ASTERISK-24304 - asterisk crashing randomly because of unistim
      channel (Reported by dhanapathy sathya)
 * ASTERISK-21721 - SIP Failed to parse multiple Supported: headers
      (Reported by Olle Johansson)
 * ASTERISK-24458 - chan_phone fails to build on big endian systems
      (Reported by Tzafrir Cohen)
 * ASTERISK-24457 - res_fax: fax gateway frames leak (Reported by
      Corey Farrell)
 * ASTERISK-24453 - manager: acl_change_sub leaks (Reported by
      Corey Farrell)
 * ASTERISK-24437 - Review implementation of ast_bridge_impart for
      leaks and document proper usage (Reported by Scott Griepentrog)
 * ASTERISK-24430 - missing letter "p" in word response in
      OriginateResponse event documentation (Reported by Dafi Ni)
 * ASTERISK-24323 - Bug in documentation AGI STREAM FILE CONTROL
      (Reported by Martin Cisárik)
 * ASTERISK-24419 - Incorrect syntax for setting language in
      configs/extensions.conf.sample (Reported by Ben Klang)
 * ASTERISK-24454 - app_queue: ao2_iterator not destroyed, causing
      leak (Reported by Corey Farrell)
 * ASTERISK-24455 - func_cdr: CDR_PROP leaks payload (Reported by
      Corey Farrell)
 * ASTERISK-24435 - Asterisk 13 with TC400P segfault (Reported by
      Marian Koniuszko)
 * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of
      SSLv3, security fix POODLE (CVE-2014-3566) (Reported by
 * ASTERISK-24122 - Documentaton for res_pjsip option use_avpf
      needs to be fixed (Reported by James Van Vleet)
 * ASTERISK-24381 - res_pjsip_sdp_rtp: Declined media streams are
      interpreted, leading to erroneous 488 rejections (Reported by
      Matt Jordan)
 * ASTERISK-24063 - [patch]Asterisk does not respect outbound proxy
      when sending qualify requests (Reported by Damian Ivereigh)
 * ASTERISK-24415 - Missing AMI VarSet events when channels inherit
      variables. (Reported by Richard Mudgett)
 * ASTERISK-24327 - bridge_native_rtp: Smart bridge operation to
      softmix sometimes fails to properly re-INVITE remotely bridged
      participants (Reported by Matt Jordan)
 * ASTERISK-24426 - CDR Batch mode: size used as time value after
      first expire (Reported by Shane Blaser)
 * ASTERISK-24312 - SIGABRT when improperly configured realtime
      pjsip  (Reported by Dafi Ni)
 * ASTERISK-23846 - Unistim multilines. Loss of voice after second
      call drops (on a second line). (Reported by Rustam Khankishyiev)
 * ASTERISK-24413 - parking/parking_tests: Crash due to assertion
      in unit tests when MoH is started on channel in holding bridge
      (Reported by Matt Jordan)
 * ASTERISK-24393 - rtptimeout=0 doesn't disable rtptimeout
      (Reported by Dmitry Melekhov)
 * ASTERISK-24321 - SIP deadlock when running automated queues
      tests (Reported by Steve Pitts)
 * ASTERISK-24392 - res_fax: fax gateway sessions leak (Reported by
      Corey Farrell)
 * ASTERISK-24237 - CDR: FRACK With PJSIP blonde transfer.
      (Reported by Richard Mudgett)
 * ASTERISK-24394 - CDR: FRACK with PJSIP directed pickup.
      (Reported by Richard Mudgett)
 * ASTERISK-18923 - res_fax_spandsp usage counter is wrong
      (Reported by Grigoriy Puzankin)
 * ASTERISK-22791 - asterisk sends Re-INVITE after receiving a BYE
      (Reported by not here)
 * ASTERISK-13797 - [patch] relax badshell tilde test (Reported by
      Tzafrir Cohen)
 * ASTERISK-24325 - res_calendar_ews: cannot be used with neon 0.30
      (Reported by Tzafrir Cohen)
 * ASTERISK-24406 - Some caller ID strings are parsed differently
      since 11.13.0 (Reported by Etienne Lessard)
 * ASTERISK-24387 - res_pjsip: rport sent from UAS MUST include the
      port that the UAC sent the request on (Reported by Matt Jordan)
 * ASTERISK-20784 - Failure to receive an ACK to a SIP Re-INVITE
      results in a SIP channel leak (Reported by NITESH BANSAL)
 * ASTERISK-15879 - [patch] Failure to receive an ACK to a SIP
      Re-INVITE results in a SIP channel leak (Reported by Torrey
 * ASTERISK-24383 - res_rtp_asterisk: Crash if no candidates
      received for component (Reported by Kevin Harwell)
 * ASTERISK-24011 - [patch]safe_asterisk tries to set ulimit -n too
      high on linux systems with lots of RAM (Reported by Michael
 * 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 - [patch] Memory leaks in chan_sip.c with
      realtime peers (Reported by ibercom)
 * ASTERISK-24350 - PJSIP shows commands prints unneeded headers
      (Reported by snuffy)
 * ASTERISK-20567 - bashism in autosupport (Reported by Tzafrir
 * ASTERISK-24357 - [fax] Out of bounds error in update_modem_bits
      (Reported by Jeremy Lainé)
 * ASTERISK-24348 - Built-in editline tab complete segfault with
      MALLOC_DEBUG (Reported by Walter Doekes)
 * ASTERISK-23768 - [patch] Asterisk man page contains a (new)
      unquoted minus sign (Reported by Jeremy Lainé)
 * ASTERISK-24295 - crash: creating out of dialog OPTIONS request
      crashes (Reported by Rogger Padilla)
 * ASTERISK-24335 - [PATCH] Asterisk incorrectly responds 503 to
      INVITE retransmissions of rejected calls (Reported by Torrey
 * 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 - [patch] Dynamic IAX2 registration stops trying
      if ever not able to resolve (Reported by David Herselman)
 * ASTERISK-24280 - Add 'rtpbindaddr' setting for chan_sip
      (Reported by Paul Belanger)
 * ASTERISK-24019 - When a Music On Hold stream starts it restarts
      at beginning of file. (Reported by Jason Richards)
 * ASTERISK-24143 - pjsip: Outbound call to WebRTC UA fails to
      transmit ACK on received 200 OK (Reported by Aleksei Kulakov)
 * ASTERISK-23997 - chan_sip: port incorrectly incremented for RTCP
      ICE candidates in SDP answer (Reported by Badalian Vyacheslav)
 * ASTERISK-24147 - ARI: channel hangup crashes asterisk process
      (Reported by Edvin Vidmar)
 * ASTERISK-23994 - res_pjsip_sdp_rtp: owner address in SDP may not
      be fully qualified domainname (Reported by Private Name)
 * ASTERISK-22252 - res_musiconhold cleanup - REF_DEBUG reload
      warnings and ref leaks (Reported by Walter Doekes)
 * ASTERISK-24178 - [patch]fromdomainport used even if not set
      (Reported by Elazar Broad)
 * ASTERISK-24229 - ARI: playback of sounds implicitly answers
      channel, preventing early media playback (Reported by Matt
 * 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 - [patch] iLBC 20 (Reported by Alexander Traud)
 * ASTERISK-26190 - [patch] SRTP: Enable AES-256 and AES-GCM.
      (Reported by Alexander Traud)
 * ASTERISK-26220 - Add support for noreturn function attributes.
      (Reported by Corey Farrell)
 * ASTERISK-22131 - Update the make dependencies script to pull,
      build, and install the correct pjproject (Reported by Matt
 * ASTERISK-25471 - [patch]Add subscribe_context to res_pjsip
      (Reported by JoshE)
 * ASTERISK-26159 - res_hep: enabled by default and information
      sent to default address (Reported by Ross Beer)
 * ASTERISK-26088 - Investigate heavy memory utilization by
      res_pjsip_pubsub (Reported by Richard Mudgett)
 * ASTERISK-25578 - [patch] SIP/SDP: No rtpmap for static RTP
      payload IDs (Reported by Alexander Traud)
 * ASTERISK-26011 - [patch]PJSIP: add "via_addr", "via_port",
      "call_id" to contacts (Reported by Alexei Gradinari)
 * ASTERISK-25965 - res_pjsip_outbound_publish: Allow multiple
      clients per configuration (Reported by Kevin Harwell)
 * ASTERISK-25994 - [patch]res_pjsip: module load priority
      (Reported by Alexei Gradinari)
 * ASTERISK-25931 - PJSIP: add "reg_server" to contacts. (Reported
      by Alexei Gradinari)
 * ASTERISK-25835 - Authentication using 'Username' field from
      Digest (Reported by Ross Beer)
 * ASTERISK-25930 - PJSIP: disable multi domain to improve realtime
      performace (Reported by Alexei Gradinari)
 * ASTERISK-25865 - Message-Account Missing From PJSIP MWI
      (Reported by Ross Beer)
 * ASTERISK-25444 - [patch]Music On Hold Warning misleading
      (Reported by Conrad de Wet)
 * ASTERISK-25846 - Gracefully deal with Absent Stasis Apps
      (Reported by Andrew Nagy)
 * ASTERISK-25791 - res_pjsip_caller_id: Lack of support for
      Anonymous <anonymous at anonymous.invalid> (Reported by Anthony
 * ASTERISK-25767 - [patch] Add check to configure for sanitizes 
      (Reported by Badalian Vyacheslav)
 * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
      core set (Reported by Rusty Newton)
 * ASTERISK-25627 - Easily Preventable Compile Warning (Reported by
      Diederik de Groot)
 * ASTERISK-25558 - [patch]chan_sip option 'notifyringing' doc fix
      and addition of 'notifyringingprio' (Reported by Ward van
 * ASTERISK-25618 - res_pjsip:  Check for readability of TLS files
      at startup (Reported by George Joseph)
 * ASTERISK-25581 - [patch]Add value reason a pause on CLI
      (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-25572 - Endpoints: Add StatsD stats for Asterisk
      endpoints (Reported by Matt Jordan)
 * ASTERISK-25571 - PJSIP: Add StatsD stats for some common PJSIP
      objects (Reported by Matt Jordan)
 * ASTERISK-25518 - taskprocessor: Add high water mark (Reported by
      Jonathan Rose)
 * ASTERISK-25495 - [patch] Prevent old-update packages on
      repository Debian systems (Reported by Rodrigo Ramirez
 * ASTERISK-25477 - pjsip show "command" like [criteria] (Reported
      by Bryant Zimmerman)
 * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
      configure (Reported by Badalian Vyacheslav)
 * ASTERISK-24870 - ARI: Subscriptions to bridges generally not
      super useful (Reported by Matt Jordan)
 * ASTERISK-25405 - [patch] CLI: core show fd: add timestamp
      (Reported by Alexander Traud)
 * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
      defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
 * ASTERISK-25256 - [patch]Post AMI VarSet to empty string events
      when Asterisk deletes a dialplan variable. (Reported by Richard
 * ASTERISK-25040 - pbx: Improve performance of reloads by making
      hint destruction more performant (Reported by Matt Jordan)
 * ASTERISK-25067 - Sorcery Caching: Implement a new caching module
      (Reported by Matt Jordan)
 * ASTERISK-25114 - res_pjsip:  Add AMI events for chan_pjsip
      contact lifecycle changes (Reported by George Joseph)
 * ASTERISK-25072 - res_pjsip_outbound_registration: line
      functionality. Additional check for using the request URI
      (Reported by Dmitriy Serov)
 * ASTERISK-24815 - [patch] Enable TLS Dual-Certificates (ECC+RSA)
      (Reported by Alexander Traud)
 * ASTERISK-25063 - [patch]add X.509 subject alternative name
      support to Asterisk TLS support (Reported by Maciej Szmigiero)
 * ASTERISK-25044 - sorcery:  Add ability to insert a new wizard
      into an object type's list (Reported by George Joseph)
 * ASTERISK-24892 - Super Awesome Company sound prompts (Reported
      by Rusty Newton)
 * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
 * ASTERISK-25049 - CLI: Enable automatic references to modules
      (Reported by Corey Farrell)
 * ASTERISK-25056 - Modules: Make ast_module_info->self available
      to auxiliary sources.  (Reported by Corey Farrell)
 * ASTERISK-25045 - vector:  Add new capabilities and unit tests
      (Reported by George Joseph)
 * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
      (Reported by Alexander Traud)
 * ASTERISK-24706 - [patch]add auto-dtmf mode for pjsip (Reported
      by yaron nahum)
 * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
      Diederik de Groot)
 * ASTERISK-25051 - Remove unneeded uses of optional_api providers.
      (Reported by Corey Farrell)
 * ASTERISK-24974 - Astobj2: Allow reference debugging to be
      enabled/disabled by config. (Reported by Corey Farrell)
 * ASTERISK-24980 - cdr_adaptive_odbc: refactor lines to
      concatenate  of columns name (Reported by Rodrigo Ramirez
 * ASTERISK-24947 - res_pjsip: Add a PJSIP resolver using core DNS
      (Reported by Joshua Colp)
 * ASTERISK-24965 - cel_pgsql - log_error string references CDR
      instead of CEL (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-24960 - Build System: Create MOD_ADD_SOURCE macro for
      module Makefiles (Reported by Corey Farrell)
 * ASTERISK-24939 - [patch]IAX make calltoken expiration time
      configurable (Reported by Y Ateya)
 * ASTERISK-24918 - pjsip: add CLI options to display global and
      system configuration (Reported by Scott Griepentrog)
 * ASTERISK-24862 - [patch] Support in-dialog OPTIONS (Reported by
      yaron nahum)
 * ASTERISK-24802 - stasis: set a channel variable on websocket
      disconnect error (Reported by Kevin Harwell)
 * ASTERISK-24133 - [patch]Please support Clang; Allow no-exec
      stacks (Reported by Jeffrey Walton)
 * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
      Couldn't find mailbox %s in context (Reported by Graham Barnett)
 * ASTERISK-24813 - asterisk.c: #if statement in listener()
      confuses code folding editors (Reported by Corey Farrell)
 * ASTERISK-24811 - asterisk-publication sorcery object does not
      use realtime (Reported by Matt Hoskins)
 * ASTERISK-24745 - [patch]Add no_answer to ARI hangup causes
      (Reported by Ben Merrills)
 * ASTERISK-24316 - For httpd server, need option to define server
      name for security purposes (Reported by Andrew Nagy)
 * ASTERISK-24671 - Missing docs for the CDR AMI Event (Reported by
      Dan Jenkins)
 * ASTERISK-24575 - [patch]Make capath work for res_pjsip (Reported
      by cloos)
 * ASTERISK-24678 - [PATCH] Added atxfer* settings to
      features.conf.sample (Reported by Niklas Larsson)
 * ASTERISK-24412 - [patch]Incomplete channel originate/continue
      handling with ARI (Reported by Nir Simionovich (GreenfieldTech -
 * ASTERISK-24351 - [patch] 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 - [patch] app_record stripping 1/4 second from
      recordings (Reported by Ben Smithurst)
 * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
      column in the cel_odbc module (Reported by Etienne Lessard)
 * ASTERISK-24128 - [Patch] Adding default dtls settings (Reported
      by Michael K.)
 * ASTERISK-24279 - Documentation: Clarify the behaviour of the CDR
      property 'unanswered' (Reported by Matt Jordan)
 * ASTERISK-23512 - Inaccurate comment in manager.conf.sample
      (Reported by Richard Miller)
 * ASTERISK-24365 - [Patch] Dialplan function to get first/head
      caller channel on queue (Reported by Kristian Høgh)
 * ASTERISK-23324 - [patch] - QLOOG commiting Japanese translated
      prompts (Reported by Kevin McCoy)
 * ASTERISK-24038 - device state: Report ONHOLD device state if
      channel driver defers device state calculation to core (Reported
      by Matt Jordan)
 * ASTERISK-24171 - [patch] Provide a manpage for the aelparse
      utility (Reported by Jeremy Lainé)
 * ASTERISK-23953 - Testsuite: Off-nominal Authenticate test
      (Reported by Matt Jordan)
 * ASTERISK-24045 - [patch]Voicemail to email at multiple email
      addresses (Reported by Jacob Barber)

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


Thank you for your continued support of Asterisk!

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