<div dir="ltr"><div class="gmail_quote"><br><br>The Asterisk Development Team has announced the release of Asterisk 12.4.0.<br>
This release is available for immediate download at<br>
<a href="http://downloads.asterisk.org/pub/telephony/asterisk" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk</a><br>
<br>
The release of Asterisk 12.4.0 resolves several issues reported by the<br>
community and would have not been possible without your participation.<br>
Thank you!<br>
<br>
The following are the issues resolved in this release:<br>
<br>
Bugs fixed in this release:<br>
-----------------------------------<br>
* ASTERISK-22551 - Session timer : UAS (Asterisk) starts counting<br>
at Invite, UAC starts counting at 200 OK. (Reported by i2045)<br>
* ASTERISK-23792 - Mutex left locked in chan_unistim.c (Reported<br>
by Peter Whisker)<br>
* ASTERISK-23582 - [patch]Inconsistent column length in *odbc<br>
(Reported by Walter Doekes)<br>
* ASTERISK-23499 - app_agent_pool: Interval hook prevents channel<br>
from being hung up (Reported by Matt Jordan)<br>
* ASTERISK-23721 - Calls to PJSIP endpoints with video enabled<br>
result in leaked RTP ports (Reported by cervajs)<br>
* ASTERISK-23803 - AMI action UpdateConfig EmptyCat clears all<br>
categories but the requested one (Reported by zvision)<br>
* ASTERISK-23718 - res_pjsip_incoming_blind_request: crash with<br>
NULL session channel (Reported by Jonathan Rose)<br>
* ASTERISK-23541 - Asterisk 12.1.0 Not respecting directmedia=no<br>
and issuing REINVITE (Reported by Justin E)<br>
* ASTERISK-23035 - ConfBridge with name longer than max (32 chars)<br>
results in several bridges with same conf_name (Reported by<br>
Iñaki Cívico)<br>
* ASTERISK-23824 - ConfBridge: Users cannot be muted via CLI or<br>
AMI when waiting to enter a conference (Reported by Matt Jordan)<br>
* ASTERISK-23683 - #includes - wildcard character in a path more<br>
than one directory deep - results in no config parsing on module<br>
reload (Reported by tootai)<br>
* ASTERISK-23827 - autoservice thread doesn't exit at shutdown<br>
(Reported by Corey Farrell)<br>
* ASTERISK-21965 - [patch] Bug-fixed version of safe_asterisk not<br>
installed over old version (Reported by Jeremy Kister)<br>
* ASTERISK-23802 - Security: Deadlock in res_pjsip_pubsub on<br>
transaction timeout (Reported by Mark Michelson)<br>
* ASTERISK-23489 - Vulnerability in res_pjsip_pubsub:<br>
unauthenticated remote crash in during MWI unsubscribe without<br>
being subscribed (Reported by John Bigelow)<br>
* ASTERISK-23609 - Security: AMI action MixMonitor allows<br>
arbitrary programs to be run (Reported by Corey Farrell)<br>
* ASTERISK-23673 - Security: DOS by consuming the number of<br>
allowed HTTP connections. (Reported by Richard Mudgett)<br>
* ASTERISK-23766 - [patch] Specify timeout for database write in<br>
SQLite (Reported by Igor Goncharovsky)<br>
* ASTERISK-23844 - Load of pbx_lua fails on sample extensions.lua<br>
with Lua 5.2 or greater due to addition of goto statement<br>
(Reported by Rusty Newton)<br>
* ASTERISK-23818 - PBX_Lua: after asterisk startup module is<br>
loaded, but dialplan not available (Reported by Dennis Guse)<br>
* ASTERISK-23834 - res_rtp_asterisk debug message gives wrong<br>
length if ICE (Reported by Richard Kenner)<br>
* ASTERISK-23922 - ao2_container nodes are inconsistent REF_DEBUG<br>
(Reported by Corey Farrell)<br>
* ASTERISK-23790 - [patch] - SIP From headers longer than 256<br>
characters result in dropped call and 'No closing bracket'<br>
warnings. (Reported by uniken1)<br>
* ASTERISK-23917 - res_http_websocket: Delay in client processing<br>
large streams of data causes disconnect and stuck socket<br>
(Reported by Matt Jordan)<br>
* ASTERISK-23908 - [patch]When using FEC error correction,<br>
asterisk tries considers negative sequence numbers as missing<br>
(Reported by Torrey Searle)<br>
* ASTERISK-23947 - ActionID missing from AMI PJSIP events<br>
(PJSIPShowEndpoints, etc.) (Reported by Mark Michelson)<br>
* ASTERISK-23921 - refcounter.py uses excessive ram for large refs<br>
files (Reported by Corey Farrell)<br>
* ASTERISK-23948 - REF_DEBUG fails to record ao2_ref against<br>
objects that were already freed (Reported by Corey Farrell)<br>
* ASTERISK-23916 - [patch]SIP/SDP fmtp line may include whitespace<br>
between attributes (Reported by Alexander Traud)<br>
* ASTERISK-23984 - Infinite loop possible in ast_careful_fwrite()<br>
(Reported by Steve Davies)<br>
* ASTERISK-23897 - [patch]Change in SETUP ACK handling (checking<br>
PI) in revision 413765 breaks working environments (Reported by<br>
Pavel Troller)<br>
* ASTERISK-24001 - res_rtp_asterisk fails to load module due to<br>
undefined symbol 'dtls_perform_handshake' when PJPROJECT is not<br>
installed (Reported by Don Fanning)<br>
<br>
Improvements made in this release:<br>
-----------------------------------<br>
* ASTERISK-23492 - Add option to safe_asterisk to disable<br>
backgrounding (Reported by Walter Doekes)<br>
* ASTERISK-23654 - Add 'pjsip reload' to default cli_aliases.conf<br>
(Reported by Rusty Newton)<br>
* ASTERISK-23811 - Improve performance of Asterisk by reducing the<br>
number of channel snapshots created (Reported by Matt Jordan)<br>
* ASTERISK-22961 - [patch] DTLS-SRTP not working with SHA-256<br>
(Reported by Jay Jideliov)<br>
* ASTERISK-23975 - Description of variables field for userEvent<br>
operation missing details. (Reported by Samuel Galarneau)<br>
* ASTERISK-23552 - http: support persistent connections (Reported<br>
by Scott Griepentrog)<br>
* ASTERISK-23939 - ARI: Allow for channel subscriptions on<br>
originate (Reported by Matt Jordan)<br>
<br>
New Features made in this release:<br>
-----------------------------------<br>
* ASTERISK-23786 - TALK_DETECT: A dialplan function that emits<br>
talking start/stop events for AMI/ARI (Reported by Matt Jordan)<br>
* ASTERISK-21443 - New SIP Channel Driver - Create a state<br>
provider for dialog-info+xml (Reported by Matt Jordan)<br>
<br>
For a full list of changes in this release, please see the ChangeLog:<br>
<br>
<a href="http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.4.0" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.4.0</a><br>
<br>
Thank you for your continued support of Asterisk!<br>
<br>
<br>--<br>
_____________________________________________________________________<br>
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