[asterisk-users] Certified Asterisk 13.13-cert1 Now Available

Asterisk Development Team asteriskteam at digium.com
Mon Feb 13 16:30:23 CST 2017

The Asterisk Development Team has announced the release of Certified Asterisk 13.13-cert1.
This release is available for immediate download at

The release of Certified Asterisk 13.13-cert1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Improvements made in this release:
 * ASTERISK-25063 - [patch]add X.509 subject alternative name
      support to Asterisk TLS support (Reported by Maciej Szmigiero)
 * ASTERISK-26558 - app_queue: add variable to know if the call is
      not answered after a queue (Reported by scgm11)
 * ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry
      (Reported by scgm11)
 * ASTERISK-26538 - codec_opus: Add sample to
      configs/samples/codecs.conf.sample (Reported by Kevin Harwell)
 * ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps', and
      'ari set debug' CLI commands (Reported by Matt Jordan)
 * ASTERISK-26418 - res_rtp_asterisk: Speed up ICE resolution by
      blacklisting host subnets that are not involved in RTP (Reported
      by Michael Walton)
 * ASTERISK-26409 - codec_opus: Update Asterisk to support the
      translation codec. (Reported by Kevin Harwell)
 * ASTERISK-26289 - Announcer channels in ConfBridges cause
      inefficiencies (Reported by Mark Michelson)
 * ASTERISK-25980 - [patch]res_fax: set FAXMODE variable to let
      dialplan know what fax transport was used (Reported by Alexei
 * 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-26059 - [patch]core: New channel variable FORWARDERNAME
      (Reported by Alexei Gradinari)
 * ASTERISK-26011 - [patch]PJSIP: add "via_addr", "via_port",
      "call_id" to contacts (Reported by Alexei Gradinari)
 * ASTERISK-26055 - [patch]res_pjsip: chatty verbose messages
      (Reported by Alexei Gradinari)
 * ASTERISK-26010 - [patch]func_odbc: single database connection
      should be optional (Reported by Alexei Gradinari)
 * 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)

Bugs fixed in this release:
 * ASTERISK-26716 - ari: Channels with pre-dial handlers cannot be
      hung up via ARI (Reported by Tom Pawelek)
 * ASTERISK-26632 - core: Possibility of a frame "imbalance"
      leading to stuck channels. (Reported by Mark Michelson)
 * ASTERISK-25951 - res_agi:  run_agi eats frames it shouldn't
      (Reported by George Joseph)
 * ASTERISK-26343 - ASTERISK-25951 causes issues for callerid
      manipulation through agi (Reported by Morten Tryfoss)
 * ASTERISK-26679 - Crash on invalid contact domain (pjsip aor)
      (Reported by Dmitriy)
 * ASTERISK-26699 - res_pjsip: Assertion when sending OPTIONS
      request  to endpoint (Reported by Ross Beer)
 * ASTERISK-26621 - app_queue: Queue application does not ring
      members with Local interface (Reported by Jonas Kellens)
 * ASTERISK-26743 - PJPROJECT: Detecting compiled max log level
      does not work. (Reported by Richard Mudgett)
 * ASTERISK-26673 - chan_pjsip: Crash when using CHANNEL dialplan
      function around masquerade (Reported by Joshua Colp)
 * ASTERISK-26672 - Crash when setting remote address on RTP
      instance (Reported by Richard Mudgett)
 * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
      bounds and missing paren issues (Reported by George Joseph)
 * ASTERISK-24499 - Need more explicit debug when PJSIP dialstring
      is invalid (Reported by Rusty Newton)
 * ASTERISK-25083 - Message.c: Message channel becomes saturated
      with frames leading to spammy log messages (Reported by Jonathan
 * ASTERISK-26433 - chan_sip: Allows To-tag checks to be bypassed,
      setting up new calls (Reported by Walter Doekes)
 * ASTERISK-26579 - codec_opus: Recursiveness when parsing fmtp
      line (Reported by Jørgen H)
 * ASTERISK-26644 - PJSIPShowRegistrationsInbound just dumps all
      aors (Reported by George Joseph)
 * ASTERISK-26490 - res_pjsip: sends 481 Call/Transaction Does Not
      Exist when transaction branch parameter contains "_" (Reported
      by Juris Breicis)
 * ASTERISK-26608 - Compile and link failures on OpenBSD (Reported
      by snuffy)
 * ASTERISK-26520 - codec_opus: Generated fmtp line has no content
      (Reported by scgm11)
 * ASTERISK-26605 - codec_opus: Spammed warning when Opus
      negotiated but codec_opus not loaded. (Reported by Richard
 * ASTERISK-26516 - pjsip: Memory corruption with possible memory
      leak. (Reported by Richard Mudgett)
 * ASTERISK-26592 - Latest libedit (3.1) defaults to unicode and
      makes asterisk CLI read garbage (Reported by George Joseph)
 * ASTERISK-26565 - chan_unistim on 11, 13, 14 placing call on hold
      temporarily locks up set (Reported by Jason)
 * ASTERISK-26575 - testsuite: Need to check PJSIP functionality
      when res_srtp is not loaded. (Reported by Joshua Colp)
 * ASTERISK-24400 - ooh323 sends wrong hangup code (Reported by
      Dmitry Melekhov)
 * ASTERISK-26555 - Multi-party Video: Fix some post Asterisk-11
      regressions (Reported by Matt Jordan)
 * ASTERISK-26412 - build:  Prepare for gcc 6.2 (Reported by George
 * ASTERISK-26509 - A few non-critical deprecation warnings when
      building on Ubuntu 16.10 (Reported by Jonathan Harris)
 * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming
      calls after 2 minutes - rtptimeout behaving badly - regression
      (Reported by Michael Keuter)
 * ASTERISK-26468 - ari: Bridge events stop working after this
      sequence of ARI calls (Reported by Daniele Pallastrelli)
 * ASTERISK-26311 - [patch] rtp_engine: Allow more than 32 dynamic
      payload types. (Reported by Alexander Traud)
 * ASTERISK-26549 - app_dial: When PickupChan() is used some
      channels may have incorrect device state (Reported by Joshua
 * ASTERISK-26541 - res_pjsip_sdp_rtp: Restrict number of formats
      to maximum (Reported by Joshua Colp)
 * ASTERISK-25070 - Fix FTBFS on Hurd (Reported by Gabriele
 * ASTERISK-26476 - chan_sip: Incorrect display option "Outbound
      reg. retry 403" in "sip show settings" (Reported by Sergey
 * ASTERISK-26537 - AMI: NewConnectedLine event is not documented
      (Reported by Etienne Lessard)
 * ASTERISK-26526 - [UBSAN] vector.h: null pointer can be passed as
      argument 2 to memcpy (Reported by Badalian Vyacheslav)
 * ASTERISK-26524 - astobj2: data_size variable is wasted space
      when AO2_DEBUG is not enabled. (Reported by Corey Farrell)
 * ASTERISK-26344 - Asterisk 13.11.0 + PJSIP crash (Reported by Ian
 * ASTERISK-26387 - Asterisk segfaults shortly after starting even
      with no active calls.  (Reported by Harley Peters)
 * ASTERISK-26514 - Super Awesome Company: Don't specify transport
      in pjsip.conf (Reported by Rusty Newton)
 * ASTERISK-26513 - tests/channels/pjsip/qualify/auth: Crashing
      enough to be a nuisance (Reported by Joshua Colp)
 * ASTERISK-26510 - pjproject_bundled uses the --strip-components
      option of tar which isn't supported in older versions (Reported
      by George Joseph)
 * ASTERISK-22480 - Embedded pjproject: build.mak contains
      hardcoded full path to version.mak (Reported by Matt Jordan)
 * ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing change
      (Reported by Bill Brigden)
 * ASTERISK-26503 - app_voicemail: Asterisk crashes when
      MailboxExists is used (Reported by Doug Lytle)
 * ASTERISK-26423 - res_pjsip_sdp_rtp: Asymmetric RTP codec can
      cause audio loss and wonkiness (Reported by Andreas Wetzel)
 * ASTERISK-26309 - [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack)
      installations. (Reported by Alexander Traud)
 * ASTERISK-26421 - Segmentation Fault with ARI originate into
      mixing bridge with 43 clients (Reported by Andrew Nagy)
 * ASTERISK-26444 - 'features show' command in CLI does not return
      prompt. (Reported by John Kiniston)
 * ASTERISK-26482 - [patch] chan_pjsip: segfault on already
      disconnected session (Reported by Alexei Gradinari)
 * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes
      File not Module (Reported by Alexander Traud)
 * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by
      Tzafrir Cohen)
 * ASTERISK-26477 - pjproject:  SEGV during SSL operations
      (Reported by George Joseph)
 * ASTERISK-26439 - chan_rtp: Crash when originating (Reported by
 * ASTERISK-17470 - [patch] - When videosupport=yes, asterisk 
      allows one end peer to send video, even though the other end
      supports only audio. (Reported by effie mouzeli)
 * ASTERISK-26462 - [patch] app_queue: While using queues with
      realtime, setting back to an empty context doesn't stop the exit
      key usage (Reported by Leandro Dardini)
 * ASTERISK-26416 - pjproject-bundled: configure fails to check for
      all required utilities (Reported by Corey Farrell)
 * ASTERISK-26466 - core: Be forgiving on external callerid that
      may be flawed so we don't drop events (Reported by Richard
 * ASTERISK-26362 - res_config_mysql:  Broken after 13.10 (Reported
      by Carlos Chavez)
 * ASTERISK-26446 - app_dial:  There's no way to override the
      hangupcause on unanswered channels (Reported by George Joseph)
 * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT
      detection triggered. (Reported by Alexander Traud)
 * ASTERISK-26453 - res_pjsip_config_wizard: Memory leak in
      module_unload (Reported by Badalian Vyacheslav)
 * ASTERISK-24311 - Populating database via Alembic fails when
      using same database for multiple schema sets (Reported by Dafi
 * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No
      Symmetric Response. (Reported by Alexander Traud)
 * ASTERISK-26426 - format_ogg_opus: remove from source (Reported
      by Kevin Harwell)
 * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when
      bindaddr=[::] (Reported by Jacek)
 * ASTERISK-25468 - Deadlock in chan_sip - core show locks shows
      do_monitor lock (Reported by Barry Flanagan)
 * ASTERISK-26397 - manager: PresenceState action crashes Asterisk
      14 (Reported by Andrew Nagy)
 * ASTERISK-26389 - res_odbc: Clean up pooling options (Reported by
      Joshua Colp)
 * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so
      instructed (Reported by Tzafrir Cohen)
 * ASTERISK-26273 - core: Won't compile when LOW_MEMORY is enabled
      (Reported by Anthony Messina)
 * ASTERISK-26352 - Astcanary dies when doing "core restart"
      (Reported by Walter Doekes)
 * ASTERISK-19867 - asterisk fails to lower its priority when
      astcanary dies (Reported by Xavier Hienne)
 * ASTERISK-26263 - SQL error when using realtime and registering
      extension / inserting into ps_contacts (Reported by Jeppe Ryskov
 * ASTERISK-26374 - res_pjsip_multihomed: Contact port is rewritten
      for connectionful protocols (Reported by Joshua Colp)
 * ASTERISK-26367 - rtp: Timestamps broken when video frame is
      across multiple RTP packets (Reported by Joshua Colp)
 * ASTERISK-26375 - res_pjsip_transport_management: Log message
      states seconds, but time value is milliseconds (Reported by
      Joshua Colp)
 * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported
      by Aaron Hamstra)
 * ASTERISK-26360 - app_queue: "queue show" output gets "failed to
      extend from 240 to 327" msgs. (Reported by Richard Mudgett)
 * ASTERISK-26358 - chan_sip: Contact is updated on re-200, but not
      on re-INVITE (Reported by Walter Doekes)
 * ASTERISK-26316 - res_pjsip_callerid: Irregular URI causes
      unexpected callerid (Reported by Kevin Harwell)
 * ASTERISK-26349 -  13.11.1 res_pjsip/pjsip_distributor.c: Request
      'REGISTER' failed (Reported by Dmitry Melekhov)
 * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets)
      (Reported by Etienne Lessard)
 * ASTERISK-26264 - res_pjsip: Crash when applying ACL from
      non-existent endpoint (Reported by nappsoft)
 * ASTERISK-26288 - followme: fails to reset config items to
      default values on reload (Reported by Tzafrir Cohen)
 * ASTERISK-23989 - [patch]SDP offer/answer fails if crypto keys
      added to non-crypto offer (Reported by Olle Johansson)
 * ASTERISK-25691 - Crash occurs when screening mode (Dial's 'p'
      argument) is enabled and callee rejects a call or hangs up.
      (Reported by Etienne Lessard)
 * ASTERISK-26331 - Crash on “core show channeltype Surrogate” in
      ast_format_cap_get_names (Reported by CGI.NET)
 * ASTERISK-26085 - app_mp3: results in timeout for streams
      (Reported by Jens Bürger)
 * ASTERISK-26282 - AEL: macro-call in Dial application, macro
      "lacks 's' extension" (Reported by chris de rock)
 * ASTERISK-26226 - pbx: Asterisk crash on AMI action
      "ShowDialplan" when there's a circular dependency between
      contexts (Reported by Etienne Lessard)
 * ASTERISK-26279 - pjproject-bundled:  Fails to compile on Debian
      6 (Reported by George Joseph)
 * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not
      cleaning up properly (Reported by Alexander Traud)
 * ASTERISK-26299 - app_queue: Queue application sometimes stops
      calling members with Local interface (Reported by Etienne
 * ASTERISK-26203 - res_fax: Deadlock when using
      FAXOPT(gateway)=yes with Local channels (Reported by Etienne
 * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking
      inversion in T.38 query option with features bridging code
      (Reported by David Brillert)
 * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with
      local channels. (Reported by Richard Mudgett)
 * ASTERISK-26269 - res_pjsip: Wrong state for aors without
      registered contacts after startup (Reported by nappsoft)
 * ASTERISK-22374 - Finish mapping the sip.conf parameters to
      res_sip.conf parameters (Reported by Matt Jordan)
 * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of
      SSLv3, security fix POODLE (CVE-2014-3566) (Reported by
 * ASTERISK-25472 - Swagger scripts are not replacing format
      variable in file brief (Reported by Corey Farrell)
 * ASTERISK-26228 - res_pjsip_sdp_rtp: G729A does not include
      annexb=no attribute. (Reported by Ali Ghavidel)
 * ASTERISK-25984 - res_odbc relies on res_odbc_transaction, but
      it's not mandatory to compile it (Reported by József Dudás)
 * ASTERISK-26305 - Asterisk 14: Two resolver unbound testsuite
      tests fail (Reported by Richard Mudgett)
 * ASTERISK-26303 - [patch] BuildSystem: ca_list_path capabilities
      not detected in PJProject. (Reported by Alexander Traud)
 * ASTERISK-25492 - ARI: Path parameters are case sensitive
      (Reported by Joshua Colp)
 * ASTERISK-26233 - pbx: Failure to remove inconsistent extension
      names (Reported by Corey Farrell)
 * ASTERISK-26164 - XMPP no longer triggers NOTIFY to device via
      chan_pjsip (Reported by Ross Beer)
 * ASTERISK-26246 - Security: Privilege escalation by AMI adding
      dialplan extensions. (Reported by Richard Mudgett)
 * ASTERISK-26267 - ast_register_atexit callbacks should be run on
      failed startup. (Reported by Corey Farrell)
 * ASTERISK-26241 - res_pjsip:  When using compact headers, rpid
      and pai are incorrectly generated (Reported by George Joseph)
 * ASTERISK-26239 - res_pjsip_logger:  An empty global/debug option
      is treated as a "match all" hostname (Reported by George Joseph)
 * ASTERISK-26238 - res_pjsip: Empty global default_from_user
      causes crash (Reported by Joshua Colp)
 * ASTERISK-25797 - app_queue: Crash when calling a queue with a
      member with a forward to an nonexistent extension (Reported by
      Etienne Lessard)
 * ASTERISK-26268 - alembic: 'auth_username' not in PJSIP
      'identify_by' enum (Reported by Joshua Colp)
 * ASTERISK-26145 - pjsip: Deadlock with suspend + masquerade +
      indicate (Reported by Ross Beer)
 * ASTERISK-26183 - alembic: error when using sqlalchemy version
      1.1.0b2 (Reported by Kevin Harwell)
 * ASTERISK-26280 - DNS lookups can block channel media paths
      (Reported by Mark Michelson)
 * ASTERISK-25217 - [patch]res_pjsip_outbound_publish.c needs a
      similar treatment for module unloading as
      res_pjsip_outbound_registration.c (Reported by Richard Mudgett)
 * ASTERISK-26265 - Errors ignored from some parts of system
      initialization. (Reported by Corey Farrell)
 * ASTERISK-26206 - [patch] res_pjsip: Use more compatible regex
      for get all (Reported by Dmitry Wagin)
 * ASTERISK-26256 - [patch] SIP/SDP origin (o=) contains brackets
      with IP6 (Reported by Alexander Traud)
 * ASTERISK-25996 - Remove "live_dangerously" requirement on
      DB(read) (Reported by Andrew Nagy)
 * ASTERISK-26148 - pjsip: Cannot compile 13.10.0-rc1:
      "libasteriskpj.so: undefined reference to..." (Reported by Hans
      van Eijsden)
 * ASTERISK-26237 - Fax is detected on regular calls. (Reported by
      Richard Mudgett)
 * ASTERISK-26227 - sqlalchemy error due to long identifier name
      (Reported by Mark Michelson)
 * 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-26199 - PJSIP: tx_data_destroy called twice (Reported
      by Scott Griepentrog)
 * ASTERISK-26166 - res_pjsip_pubsub: Crash when decrementing
      reference count of message (Reported by Ross Beer)
 * ASTERISK-26174 - res_pjsip: Crash when freeing cloned message in
      distributor (Reported by Ross Beer)
 * ASTERISK-26216 - res_fax: Deadlock when detect fax while channel
      executing Playback (Reported by Richard Mudgett)
 * 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-26200 - [patch] res_pjsip_mwi: improve realtime
      performance - remove unneeded check on endpoint's contacts.
      (Reported by Alexei Gradinari)
 * 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-25659 - res_rtp_asterisk: ECDH not negotiated causing
      DTLS failure occurred on RTP instance (Reported by Edwin
 * ASTERISK-26191 - threadpool: Leak on duplicate taskprocessor for
      ast_threadpool_serializer_group (Reported by Corey Farrell)
 * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf.
      (Reported by Alexander Traud)
 * ASTERISK-26160 - pjsip: Updated->Reachable during qualify
      (Reported by Matt Jordan)
 * 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-26177 - func_odbc: Database handle is kept when it
      should be released (Reported by Leandro Dardini)
 * 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-26180 - PJSIP: provide valid tcp nodelay option for
      reuse (Reported by Scott Griepentrog)
 * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported
      by Joshua Colp)
 * ASTERISK-26172 - res_sorcery_realtime: fix bug when successful
      sql UPDATE is treated as failed if there is no affected rows.
      (Reported by Alexei Gradinari)
 * 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-26061 - [patch] res_pjsip: improve realtime performance
      - remove updating all endpoints status on startup (Reported by
      Alexei Gradinari)
 * ASTERISK-26129 - res_rtp_asterisk: Memory leak of CERT bio in
      DTLS implementation (Reported by Torrey Searle)
 * 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-26069 - Asterisk truncates To: header, dropping the
      closing '>' (Reported by Vasil Kolev)
 * 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-26092 - [Segfault] in res_rtp_asterisk.c:4268 after
      Remotely bridged channels (Reported by Niklas Larsson)
 * ASTERISK-26096 - res_hep: Crash when configuration file is
      missing (Reported by Niklas Larsson)
 * ASTERISK-26089 - Invalid security events during boot using PJSIP
      Realtime (Reported by Scott Griepentrog)
 * ASTERISK-26074 - res_odbc: Deadlock within UnixODBC (Reported by
      Ross Beer)
 * ASTERISK-26054 - Asterisk crashes (core dump) (Reported by B.
 * 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-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-26008 - app_followme does not delete recorded name
      prompt (Reported by Tzafrir Cohen)
 * 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-25538 - [patch]Missing PID in syslog logger messages
      (Reported by Javier Acosta)
 * 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-25993 - pjproject: Allow bundling to not require
      everything it does (Reported by Joshua Colp)
 * ASTERISK-25956 - Compilation error in conditionally compiled
      code in config_options.c (Reported by Chris Trobridge)
 * ASTERISK-25998 - file: Crash when using nativeformats (Reported
      by Joshua Colp)
 * ASTERISK-25826 - PJSIP / Sorcery slow load from realtime
      (Reported by Ross Beer)
 * ASTERISK-25982 - [patch]res_fax/t38_gateway: Peer V.21 session
      is created on wrong channel (Reported by Alexei Gradinari)
 * 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-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-25950 - [patch]SIP channel does not send PeerStatus
      events for autocreated peers (Reported by Kirill Katsnelson)
 * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions
      are case sensitive to QueueName (Reported by Javier Acosta)
 * ASTERISK-25927 - Removed option "registertrying" is still
      documented in sip.conf.sample (Reported by Etienne Lessard)
 * ASTERISK-25948 - ast_pthread_mutex_lock calling
      ast_reentrancy_lock with lt=0x0 (Reported by Diederik de Groot)
 * 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-24927 - app_voicemail (IMAP support) function
      save_to_folder: creates wrong folder (Reported by Alexei
 * ASTERISK-25914 - PJSIP: failed registration with wrong codec
      name on allow/disallow (Reported by Alexei Gradinari)
 * 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-25899 - IMAP access FATAL error: Out of memory
      (Reported by Alexei Gradinari)
 * 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-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-25642 - res_rtp_asterisk: SRTCP broken with DTLS  - bad
      video is one of the consequences (Reported by Stefan Engström)
 * 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-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-25612 - Configuration parser handles unsigned integers
      as signed integers (Reported by Gianluca Merlo)
 * ASTERISK-25825 - Crashes during shutdown when running CLI
      commands (Reported by Mark Michelson)
 * 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-21301 - ERROR and failure to resolve socket address due
      to whitespace after port number in SIP Via header (Reported by
      Martin Vit)
 * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
      data corruption (Reported by Gianluca Merlo)

New Features made in this release:
 * ASTERISK-26630 - Make logging PJPROJECT messages a bit easier
      (Reported by Richard Mudgett)
 * ASTERISK-26595 - ARI: Add the ability to control the source of
      video in a multi-party mixing bridge (Reported by Matt Jordan)
 * ASTERISK-26470 - ARI: Add an 'asterisk_id' field to outgoing
      events (Reported by Matt Jordan)
 * ASTERISK-26277 - Add dialplan function
      PJSIP_SEND_SESSION_REFRESH that sends a session refresh to
      update formats on a channel after session establishment
      (Reported by Matt Jordan)
 * ASTERISK-25904 - PJSIP: add contact.updated event (Reported by
      Alexei Gradinari)
 * ASTERISK-25900 - PJSIP Endpoint IP Access Controls (Reported by
      Alexei Gradinari)
 * ASTERISK-25989 - apps/confbridge: add regcontext feature
      (Reported by Jaco Kroon)
 * ASTERISK-25903 - PJSIP AMI Event ContactStatus: add Useragent
      and RegExpire (Reported by Alexei Gradinari)
 * ASTERISK-25901 - Add transport for outbound PUBLISH (Reported by
      Alexei Gradinari)

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


Thank you for your continued support of Asterisk!

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