[asterisk-announce] Asterisk 13.4.0 Now Available

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The Asterisk Development Team has announced the release of Asterisk 13.4.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.4.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

New Features made in this release:
-----------------------------------
 * ASTERISK-24922 - ARI: Add the ability to intercept hold and
      raise an event (Reported by Matt Jordan)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-25112 - Logger: Configuration settings are not reset to
      default during reload. (Reported by Corey Farrell)
 * ASTERISK-24944 - main/audiohook.c change prevents G722 call
      recording (Reported by Ronald Raikes)
 * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
      or more digits (Reported by Makoto Dei)
 * ASTERISK-25086 - [patch]PJSIP crashes if endpoint missing in
      Dial() (Reported by snuffy)
 * ASTERISK-25089 - res_pjsip_config_wizard: Variable specified in
      templates aren't being processed correctly (Reported by George
      Joseph)
 * ASTERISK-25090 - CLI core show channel truncates cdr variables
      (Reported by snuffy)
 * ASTERISK-25085 - [patch]Potential crash after unload of
      func_periodic_hook or test_message (Reported by Corey Farrell)
 * ASTERISK-25083 - Message.c: Message channel becomes saturated
      with frames leading to spammy log messages (Reported by Jonathan
      Rose)
 * ASTERISK-25082 - Asterisk deletes message after doing a playback
      of an INBOX message using ast_vm_play when the Old folder is
      full for that mailbox. (Reported by Jonathan Rose)
 * ASTERISK-25041 - [patch]Broken column type checking in
      res_config_mysql addon (Reported by Alexandre Fournier)
 * ASTERISK-21893 - Segfault after call hangup, in
      ast_channel_hangupcause_set, at channel_internal_api.c (Reported
      by Alexandr Gordeev)
 * ASTERISK-25074 - Regression: Recent clang-related change broke
      cross compiling of Asterisk (Reported by Sebastian Kemper)
 * ASTERISK-25042 - asterisk.conf options override command-line
      options. (Reported by Corey Farrell)
 * ASTERISK-24442 - Outgoing call files don't work properly when
      set in the future (Reported by tootai)
 * ASTERISK-25057 - res_pjsip_pubsub: Crash in send_notify due to
      invalid root pointer in sub_tree (Reported by Matt Jordan)
 * ASTERISK-24938 - ARI Snoop Channel results in excessive
      escalating CPU usage (Reported by George Ladoff)
 * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
      ignore ISDN RESTART requests. (Reported by Richard Mudgett)
 * ASTERISK-25003 - Asterisk crashes on attended transfer (using
      feature) (Reported by Artem Volodin)
 * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
      contain waiting time (Reported by Etienne Lessard)
 * ASTERISK-25027 - Build System: Many ARI modules are missing
      dependencies. (Reported by Corey Farrell)
 * ASTERISK-25061 - pbx_config: Register manager actions with
      module version of macro. (Reported by Corey Farrell)
 * ASTERISK-25025 - Periodic crashes (in
      ast_channel_snapshot_create at stasis_channels.c) with Certified
      Asterisk 13. (Reported by Chet Stevens)
 * ASTERISK-25053 - Unit test category /main/presence missing
      trailing slash. (Reported by Corey Farrell)
 * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
      not respected, failover between DSNs doesn't work (Reported by
      JoshE)
 * ASTERISK-25054 - Formats interface's cannot be unregistered,
      needs to hold modules until shutdown. (Reported by Corey
      Farrell)
 * ASTERISK-24896 - [patch] Using force black background leads to
      colours not being reset (Reported by dant)
 * ASTERISK-25033 - Asterisk 13 (branch head) won't compile without
      PJSip (Reported by Peter Whisker)
 * ASTERISK-25028 - Build System: Unneeded defines in
      asterisk/buildopts.h (Reported by Corey Farrell)
 * ASTERISK-25048 - Astobj2: Initialization order wrong when both
      refdebug and AO2_DEBUG are both enabled. (Reported by Corey
      Farrell)
 * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
      cause code 44 after some time. (Reported by Denis Alberto
      Martinez)
 * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
      (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-25037 - res_pjsip_outbound_registration: Potential
      crash in off-nominal failure case when sending message (Reported
      by Joshua Colp)
 * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
      (Reported by Steve Davies)
 * ASTERISK-22790 - check_modem_rate() may return incorrect rate
      for V.27 (Reported by not here)
 * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
      to minrate=2400, then res_fax refuse to load (Reported by David
      Brillert)
 * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
      which is disallowed in res_fax's check_modem_rate (Reported by
      Matt Jordan)
 * ASTERISK-24996 - chan_pjsip: Creating Channel Causes Asterisk to
      Crash When Duplicate AOR Sections Exist in pjsip.conf (Reported
      by Ashley Sanders)
 * ASTERISK-25020 - Mismatched response to outgoing REGISTER
      request (Reported by Mark Michelson)
 * ASTERISK-25018 - pjsip show endpoints crashes asterisk when
      qualified aors present (Reported by Ivan Poddubny)
 * ASTERISK-24749 - ConfBridge: Wrong language on playing
      conf-hasjoin and conf-hasleft when played to bridge (Reported by
      Philippe Bolduc)
 * ASTERISK-24845 - pjsip send notify not working with Cisco phone
      (Reported by Carl Fortin)
 * ASTERISK-25004 - Crash in authenticated reinvite after
      originated T.38 FAX (Reported by Mark Michelson)
 * ASTERISK-24999 - PJSIP crashes with malformed contact line
      (Reported by snuffy)
 * ASTERISK-24998 - res_corosync:  res_corosync tries to load even
      if res_corosync.conf is missing (Reported by George Joseph)
 * ASTERISK-24997 - Astobj2: Some callers of __adjust_lock do not
      pre-check the object (Reported by Corey Farrell)
 * ASTERISK-24982 - res_pjsip_mwi: Unsolicited MWI NOTIFY only sent
      on mailbox changes (Reported by Joshua Colp)
 * ASTERISK-24991 - Check for ao2_alloc failure in
      __ast_channel_internal_alloc (Reported by Corey Farrell)
 * ASTERISK-24895 - After hangup on the side of the ISDN network no
      HangupRequest event comes for the dahdi channel. (Reported by
      Andrew Zherdin)
 * ASTERISK-24977 - Contacts that don't use qualify are being
      marked as unavailable (Reported by George Joseph)
 * ASTERISK-24774 - Segfault in ast_context_destroy with
      extensions.ael and extensions.conf (Reported by Corey Farrell)
 * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when
      channels have multiple native formats (Reported by Matt Jordan)
 * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
      to Fail (Reported by Ashley Sanders)
 * ASTERISK-24958 - Forwarding loop detection inhibits certain
      desirable scenarios (Reported by Mark Michelson)
 * ASTERISK-24863 - res_pjsip: No endpoint events raised via AMI
      when contacts cannot be reached/qualified (Reported by Dmitriy
      Serov)
 * ASTERISK-24869 - Asterisk segfaults on DAHDI attended transfer
      due to application (appl) being NULL on unbridged channel
      (Reported by viniciusfontes)
 * ASTERISK-24970 - Crash in res_pjsip_pubsub handling of failed
      notify (Reported by Scott Griepentrog)
 * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
      (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-24954 - Git migration: Asterisk version numbers are
      incompatible with the Test Suite (Reported by Matt Jordan)
 * ASTERISK-17608 - func_aes.so cannot be loaded if res_crypto /
      openssl not compiled (Reported by Warren Selby)
 * ASTERISK-24928 - [patch]t38_udptl_maxdatagram in pjsip.conf not
      honored (Reported by Juergen Spies)
 * ASTERISK-24835 - Early Media Not working with Chan SIP and
      Asterisk 13 (Reported by Andrew Nagy)
 * ASTERISK-21777 - Asterisk tries to transcode video instead of
      audio (Reported by Nick Ruggles)
 * ASTERISK-24380 - core: Native formats are set to h264 with
      certain audio/video codec configuration, resulting in path
      translation WARNINGs (Reported by Matt Jordan)
 * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
      into account (Reported by Frederic Van Espen)
 * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
      short (Reported by Y Ateya)
 * ASTERISK-24935 - res_pjsip_phoneprov_provider: Fix leaked
      OBJ_MULTIPLE iterator. (Reported by Corey Farrell)
 * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
      (Reported by Vadim)
 * ASTERISK-24933 - T38 fails negotiation (Reported by Jonathan
      Rose)
 * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
      byte prefix bug (Reported by Matt Jordan)
 * ASTERISK-21211 - chan_iax2 - unprotected access of
      iaxs[peer->callno] potentially results in segfault (Reported by
      Jaco Kroon)
 * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
      (Reported by Christoph Timm)
 * ASTERISK-24782 - StasisEnd event not present for channel that
      was swapped out for another after completing attended transfer
      (Reported by John Bigelow)
 * ASTERISK-24910 - "timer=no" and "timer=required" settings in
      pjsip.conf fail (Reported by Ray Crumrine)
 * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
      (Reported by Jeffrey C. Ollie)
 * ASTERISK-24914 - Division by zero in file.c when playback of
      voicemail with video as h264 (Reported by Marcello Ceschia)
 * ASTERISK-24899 - Parking fall-through behavior different in 13
      (Reported by Malcolm Davenport)
 * ASTERISK-24937 - [patch]res_pjsip_messaging: Messages may be
      sent out of order (Reported by Mark Michelson)
 * ASTERISK-24920 - Asterisk handles duplicate SIP requests as if
      they were each a new request (Reported by Mark Michelson)
 * ASTERISK-24857 - [patch] "timing test", pjsip incoming/outgoing
      calls, voicemail prompts and recordings all fail when using the
      kqueue timer source on FreeBSD 10.x (Reported by Justin T.
      Gibbs)
 * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
      detection in ast_malloc (Reported by Timo Ter??s)
 * ASTERISK-24142 - CCSS: crash during shutdown due to device
      lookup in destroyed container (Reported by David Brillert)
 * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
      core restart now (Reported by Peter Katzmann)
 * ASTERISK-24805 - [patch] - ASAN: Race condition
      (heap-use-after-free) on asterisk closing (Reported by Badalian
      Vyacheslav)
 * ASTERISK-24881 - ast_register_atexit should only be used when
      absolutely needed (Reported by Corey Farrell)
 * ASTERISK-24731 - res_pjsip_session cannot be unloaded (Reported
      by Corey Farrell)
 * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
      (Reported by Kevin Harwell)
 * ASTERISK-14233 - [patch] Buddies are always auto-registered when
      processing the roster (Reported by Simon Arlott)
 * ASTERISK-24780 - [patch] - Buddies are always auto-registered
      when processing the roster (Reported by Simon Arlott)
 * ASTERISK-24781 - PJSIP: Unnecessary 180 Ringing messages sent
      with undesireabe consequences. (Reported by Richard Mudgett)

Improvements made in this release:
-----------------------------------
 * ASTERISK-25044 - sorcery:  Add ability to insert a new wizard
      into an object type's list (Reported by George Joseph)
 * ASTERISK-24892 - Super Awesome Company sound prompts (Reported
      by Rusty Newton)
 * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
      Hjelm)
 * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
      (Reported by Alexander Traud)
 * ASTERISK-25045 - vector:  Add new capabilities and unit tests
      (Reported by George Joseph)
 * ASTERISK-24706 - [patch]add auto-dtmf mode for pjsip (Reported
      by yaron nahum)
 * ASTERISK-25051 - Remove unneeded uses of optional_api providers.
      (Reported by Corey Farrell)
 * ASTERISK-25040 - pbx: Improve performance of reloads by making
      hint destruction more performant (Reported by Matt Jordan)
 * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
      Diederik de Groot)
 * ASTERISK-24949 - res_pjsip_outbound_registration: Backport line
      functionality (Reported by Joshua Colp)
 * ASTERISK-24965 - cel_pgsql - log_error string references CDR
      instead of CEL (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-24918 - pjsip: add CLI options to display global and
      system configuration (Reported by Scott Griepentrog)
 * ASTERISK-24862 - [patch] Support in-dialog OPTIONS (Reported by
      yaron nahum)
 * ASTERISK-24802 - stasis: set a channel variable on websocket
      disconnect error (Reported by Kevin Harwell)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.4.0

Thank you for your continued support of Asterisk!



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