[asterisk-dev] Asterisk 12.4.0-rc1 Now Available

Asterisk Development Team asteriskteam at digium.com
Tue Jul 8 15:24:04 CDT 2014 

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

The release of Asterisk 12.4.0-rc1 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 candidate:

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-23814 - No call started after peer dialed (Reported by
      Igor Goncharovsky)
 * 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 candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.4.0-rc1

Thank you for your continued support of Asterisk!

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