[asterisk-dev] Asterisk 12.4.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the release of Asterisk 12.4.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 12.4.0 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:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-22551 - Session timer : UAS (Asterisk) starts counting
      at Invite, UAC starts counting at 200 OK. (Reported by i2045)
 * ASTERISK-23792 - Mutex left locked in chan_unistim.c (Reported
      by Peter Whisker)
 * ASTERISK-23582 - [patch]Inconsistent column length in *odbc
      (Reported by Walter Doekes)
 * ASTERISK-23499 - app_agent_pool: Interval hook prevents channel
      from being hung up (Reported by Matt Jordan)
 * ASTERISK-23721 - Calls to PJSIP endpoints with video enabled
      result in leaked RTP ports (Reported by cervajs)
 * ASTERISK-23803 - AMI action UpdateConfig EmptyCat clears all
      categories but the requested one (Reported by zvision)
 * ASTERISK-23718 - res_pjsip_incoming_blind_request: crash with
      NULL session channel (Reported by Jonathan Rose)
 * ASTERISK-23541 - Asterisk 12.1.0 Not respecting directmedia=no
      and issuing REINVITE (Reported by Justin E)
 * ASTERISK-23035 - ConfBridge with name longer than max (32 chars)
      results in several bridges with same conf_name (Reported by
      Iñaki Cívico)
 * ASTERISK-23824 - ConfBridge: Users cannot be muted via CLI or
      AMI when waiting to enter a conference (Reported by Matt Jordan)
 * ASTERISK-23683 - #includes - wildcard character in a path more
      than one directory deep - results in no config parsing on module
      reload (Reported by tootai)
 * ASTERISK-23827 - autoservice thread doesn't exit at shutdown
      (Reported by Corey Farrell)
 * ASTERISK-21965 - [patch] Bug-fixed version of safe_asterisk not
      installed over old version (Reported by Jeremy Kister)
 * ASTERISK-23802 - Security: Deadlock in res_pjsip_pubsub on
      transaction timeout (Reported by Mark Michelson)
 * ASTERISK-23489 - Vulnerability in res_pjsip_pubsub:
      unauthenticated remote crash in during MWI unsubscribe without
      being subscribed (Reported by John Bigelow)
 * ASTERISK-23609 - Security: AMI action MixMonitor allows
      arbitrary programs to be run (Reported by Corey Farrell)
 * ASTERISK-23673 - Security: DOS by consuming the number of
      allowed HTTP connections. (Reported by Richard Mudgett)
 * ASTERISK-23766 - [patch] Specify timeout for database write in
      SQLite (Reported by Igor Goncharovsky)
 * ASTERISK-23844 - Load of pbx_lua fails on sample extensions.lua
      with Lua 5.2 or greater due to addition of goto statement
      (Reported by Rusty Newton)
 * ASTERISK-23818 - PBX_Lua: after asterisk startup module is
      loaded, but dialplan not available (Reported by Dennis Guse)
 * ASTERISK-23834 - res_rtp_asterisk debug message gives wrong
      length if ICE (Reported by Richard Kenner)
 * ASTERISK-23922 - ao2_container nodes are inconsistent REF_DEBUG
      (Reported by Corey Farrell)
 * ASTERISK-23790 - [patch] - SIP From headers longer than 256
      characters result in dropped call and 'No closing bracket'
      warnings. (Reported by uniken1)
 * ASTERISK-23917 - res_http_websocket: Delay in client processing
      large streams of data causes disconnect and stuck socket
      (Reported by Matt Jordan)
 * ASTERISK-23908 - [patch]When using FEC error correction,
      asterisk tries considers negative sequence numbers as missing
      (Reported by Torrey Searle)
 * ASTERISK-23947 - ActionID missing from AMI PJSIP events
      (PJSIPShowEndpoints, etc.) (Reported by Mark Michelson)
 * ASTERISK-23921 - refcounter.py uses excessive ram for large refs
      files  (Reported by Corey Farrell)
 * ASTERISK-23948 - REF_DEBUG fails to record ao2_ref against
      objects that were already freed (Reported by Corey Farrell)
 * ASTERISK-23916 - [patch]SIP/SDP fmtp line may include whitespace
      between attributes (Reported by Alexander Traud)
 * ASTERISK-23984 - Infinite loop possible in ast_careful_fwrite()
      (Reported by Steve Davies)
 * ASTERISK-23897 - [patch]Change in SETUP ACK handling (checking
      PI) in revision 413765 breaks working environments (Reported by
      Pavel Troller)
 * ASTERISK-24001 - res_rtp_asterisk fails to load module due to
      undefined symbol 'dtls_perform_handshake' when PJPROJECT is not
      installed (Reported by Don Fanning)

Improvements made in this release:
-----------------------------------
 * ASTERISK-23492 - Add option to safe_asterisk to disable
      backgrounding (Reported by Walter Doekes)
 * ASTERISK-23654 - Add 'pjsip reload' to default cli_aliases.conf
      (Reported by Rusty Newton)
 * ASTERISK-23811 - Improve performance of Asterisk by reducing the
      number of channel snapshots created (Reported by Matt Jordan)
 * ASTERISK-22961 - [patch] DTLS-SRTP not working with SHA-256
      (Reported by Jay Jideliov)
 * ASTERISK-23975 - Description of variables field for userEvent
      operation missing details. (Reported by Samuel Galarneau)
 * ASTERISK-23552 - http: support persistent connections (Reported
      by Scott Griepentrog)
 * ASTERISK-23939 - ARI: Allow for channel subscriptions on
      originate (Reported by Matt Jordan)

New Features made in this release:
-----------------------------------
 * ASTERISK-23786 - TALK_DETECT: A dialplan function that emits
      talking start/stop events for AMI/ARI (Reported by Matt Jordan)
 * ASTERISK-21443 - New SIP Channel Driver - Create a state
      provider for dialog-info+xml (Reported by Matt Jordan)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.4.0

Thank you for your continued support of Asterisk!