<div dir="ltr"><div class="gmail_quote"><br>The Asterisk Development Team has announced the release of Asterisk 13.3.0.<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 13.3.0 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>
New Features made in this release:<br>
-----------------------------------<br>
* ASTERISK-24703 - ARI: Add the ability to "transfer" (redirect) a<br>
channel (Reported by Matt Jordan)<br>
* ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation<br>
(Reported by Dwayne Hubbard)<br>
<br>
Bugs fixed in this release:<br>
-----------------------------------<br>
* ASTERISK-24616 - Crash in res_format_attr_h264 due to invalid<br>
string copy (Reported by Yura Kocyuba)<br>
* ASTERISK-24748 - res_pjsip: If wizards explicitly configured in<br>
sorcery.conf false ERROR messages may occur (Reported by Joshua<br>
Colp)<br>
* ASTERISK-24769 - res_pjsip_sdp_rtp: Local ICE candidates leaked<br>
(Reported by Matt Jordan)<br>
* ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in<br>
res_odbc (Reported by ibercom)<br>
* ASTERISK-24479 - Enable REF_DEBUG for module references<br>
(Reported by Corey Farrell)<br>
* ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to<br>
fully disconnect underlying socket, leading to events being<br>
dropped with no additional information (Reported by Matt Jordan)<br>
* ASTERISK-24772 - ODBC error in realtime sippeers when device<br>
unregisters under MariaDB (Reported by Richard Miller)<br>
* ASTERISK-24752 - Crash in bridge_manager_service_req when bridge<br>
is destroyed by ARI during shutdown (Reported by Richard<br>
Mudgett)<br>
* ASTERISK-24741 - dtls_handler causes Asterisk to crash (Reported<br>
by Zane Conkle)<br>
* ASTERISK-24015 - app_transfer fails with PJSIP channels<br>
(Reported by Private Name)<br>
* ASTERISK-24727 - PJSIP: Crash experienced during multi-Asterisk<br>
transfer scenario. (Reported by Mark Michelson)<br>
* ASTERISK-24771 - ${CHANNEL(pjsip)} - segfault (Reported by<br>
Niklas Larsson)<br>
* ASTERISK-24716 - Improve pjsip log messages for presence<br>
subscription failure (Reported by Rusty Newton)<br>
* ASTERISK-24612 - res_pjsip: No information if a required sorcery<br>
wizard is not loaded (Reported by Joshua Colp)<br>
* ASTERISK-24768 - res_timing_pthread: file descriptor leak<br>
(Reported by Matthias Urlichs)<br>
* ASTERISK-24685 - "pjsip show version" CLI command (Reported by<br>
Joshua Colp)<br>
* ASTERISK-24632 - install_prereq script installs pjproject<br>
without IPv6 support (Reported by Rusty Newton)<br>
* ASTERISK-24085 - Documentation - We should remove or further<br>
document the 'contact' section in pjsip.conf (Reported by Rusty<br>
Newton)<br>
* ASTERISK-24791 - Crash in ast_rtcp_write_report (Reported by<br>
JoshE)<br>
* ASTERISK-24700 - CRASH: NULL channel is being passed to<br>
ast_bridge_transfer_attended() (Reported by Zane Conkle)<br>
* ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove<br>
(Reported by Corey Farrell)<br>
* ASTERISK-24799 - [patch] make fails with undefined reference to<br>
SSLv3_client_method (Reported by Alexander Traud)<br>
* ASTERISK-22670 - Asterisk crashes when processing ISDN AoC<br>
Events (Reported by klaus3000)<br>
* ASTERISK-24689 - Segfault on hangup after outgoing PRI-Euroisdn<br>
call (Reported by Marcel Manz)<br>
* ASTERISK-24740 - [patch]Segmentation fault on aoc-e event<br>
(Reported by Panos Gkikakis)<br>
* ASTERISK-24787 - [patch] - Microsoft exchange incompatibility<br>
for playing back messages stored in IMAP - play_message: No<br>
origtime (Reported by Graham Barnett)<br>
* ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc<br>
OSX with 64 bit integers (Reported by Corey Farrell)<br>
* ASTERISK-24796 - Codecs and bucket schema's prevent module<br>
unload (Reported by Corey Farrell)<br>
* ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML<br>
(Reported by Ashley Sanders)<br>
* ASTERISK-24499 - Need more explicit debug when PJSIP dialstring<br>
is invalid (Reported by Rusty Newton)<br>
* ASTERISK-24785 - 'Expires' header missing from 200 OK on<br>
REGISTER (Reported by Ross Beer)<br>
* ASTERISK-24677 - ARI GET variable on channel provides unhelpful<br>
response on non-existent variable (Reported by Joshua Colp)<br>
* ASTERISK-24797 - bridge_softmix: G.729 codec license held<br>
(Reported by Kevin Harwell)<br>
* ASTERISK-24812 - ARI: Creating channels through /channels<br>
resource always uses SLIN, which results in unneeded transcoding<br>
(Reported by Matt Jordan)<br>
* ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid<br>
thread ID being passed to pthread_kill (Reported by JoshE)<br>
* ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime<br>
fail (Reported by Terry Wilson)<br>
* ASTERISK-23214 - chan_sip WARNING message 'We are requesting<br>
SRTP for audio, but they responded without it' is ambiguous and<br>
wrong in some cases (Reported by Rusty Newton)<br>
* ASTERISK-15434 - [patch] When ast_pbx_start failed, both an<br>
error response and BYE are sent to the caller (Reported by<br>
Makoto Dei)<br>
* ASTERISK-18105 - most of asterisk modules are unbuildable in<br>
cygwin environment (Reported by feyfre)<br>
* ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)<br>
* ASTERISK-24751 - Integer values in json payload to ARI cause<br>
asterisk to crash (Reported by jeffrey putnam)<br>
* ASTERISK-24838 - chan_sip: Locking inversion occurs when<br>
building a peer causes a peer poke during request handling<br>
(Reported by Richard Mudgett)<br>
* ASTERISK-24825 - Caller ID not recognized using<br>
Centrex/Distinctive dialing (Reported by Richard Mudgett)<br>
* ASTERISK-24830 - res_rtp_asterisk.c checks USE_PJPROJECT not<br>
HAVE_PJPROJECT (Reported by Stefan Engström)<br>
* ASTERISK-24840 - res_pjsip: conflicting endpoint identifiers<br>
(Reported by Kevin Harwell)<br>
* ASTERISK-24755 - Asterisk sends unexpected early BYE to<br>
transferrer during attended transfer when using a Stasis bridge<br>
(Reported by John Bigelow)<br>
* ASTERISK-24739 - [patch] - Out of files -- call fails --<br>
numerous files with inodes from under /usr/share/zoneinfo,<br>
mostly posixrules (Reported by Ed Hynan)<br>
* ASTERISK-23390 - NewExten Event with application AGI shows up<br>
before and after AGI runs (Reported by Benjamin Keith Ford)<br>
* ASTERISK-24786 - [patch] - Asterisk terminates when playing a<br>
voicemail stored in LDAP (Reported by Graham Barnett)<br>
* ASTERISK-24808 - res_config_odbc: Improper escaping of<br>
backslashes occurs with MySQL (Reported by Javier Acosta)<br>
* ASTERISK-24807 - Missing mandatory field Max-Forwards (Reported<br>
by Anatoli)<br>
* ASTERISK-20850 - [patch]Nested functions aren't portable.<br>
Adapting RAII_VAR to use clang/llvm blocks to get the<br>
same/similar functionality. (Reported by Diederik de Groot)<br>
* ASTERISK-24872 - [patch] AMI PJSIPShowEndpoint closes AMI<br>
connection on error (Reported by Dmitriy Serov)<br>
* ASTERISK-19470 - Documentation on app_amd is incorrect (Reported<br>
by Frank DiGennaro)<br>
* ASTERISK-21038 - Bad command completion of "core set debug<br>
channel" (Reported by Richard Kenner)<br>
* ASTERISK-18708 - func_curl hangs channel under load (Reported by<br>
Dave Cabot)<br>
* ASTERISK-16779 - Cannot disallow unknown format '' (Reported by<br>
Atis Lezdins)<br>
* ASTERISK-24876 - Investigate reference leaks from<br>
tests/channels/local/local_optimize_away (Reported by Corey<br>
Farrell)<br>
* ASTERISK-24882 - chan_sip: Improve usage of REF_DEBUG (Reported<br>
by Corey Farrell)<br>
* ASTERISK-24817 - init_logger_chain: unreachable code block<br>
(Reported by Corey Farrell)<br>
* ASTERISK-24880 - [patch]Compilation under OpenBSD (Reported by<br>
snuffy)<br>
* ASTERISK-24879 - [patch]Compilation fails due to 64bit time<br>
under OpenBSD (Reported by snuffy)<br>
<br>
Improvements made in this release:<br>
-----------------------------------<br>
* ASTERISK-24745 - [patch]Add no_answer to ARI hangup causes<br>
(Reported by Ben Merrills)<br>
* ASTERISK-24811 - asterisk-publication sorcery object does not<br>
use realtime (Reported by Matt Hoskins)<br>
* ASTERISK-24790 - Reduce spurious noise in logs from voicemail -<br>
Couldn't find mailbox %s in context (Reported by Graham Barnett)<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-13.3.0" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.0</a><br>
<br>
Thank you for your continued support of Asterisk!<br>
<br>
<br>--<br>
_____________________________________________________________________<br>
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