<div dir="ltr"><div class="gmail_quote"><br>The Asterisk Development Team has announced the release of Asterisk 12.7.0.<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 12.7.0 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-24339 - Swagger API Docs have incorrect basePath<br>
(Reported by Bradley Watkins)<br>
* ASTERISK-24348 - Built-in editline tab complete segfault with<br>
MALLOC_DEBUG (Reported by Walter Doekes)<br>
* ASTERISK-24335 - [PATCH] Asterisk incorrectly responds 503 to<br>
INVITE retransmissions of rejected calls (Reported by Torrey<br>
Searle)<br>
* ASTERISK-24295 - crash: creating out of dialog OPTIONS request<br>
crashes (Reported by Rogger Padilla)<br>
* ASTERISK-23768 - [patch] Asterisk man page contains a (new)<br>
unquoted minus sign (Reported by Jeremy Lainé)<br>
* ASTERISK-24357 - [fax] Out of bounds error in update_modem_bits<br>
(Reported by Jeremy Lainé)<br>
* ASTERISK-20567 - bashism in autosupport (Reported by Tzafrir<br>
Cohen)<br>
* ASTERISK-24350 - PJSIP shows commands prints unneeded headers<br>
(Reported by snuffy)<br>
* ASTERISK-22945 - [patch] Memory leaks in chan_sip.c with<br>
realtime peers (Reported by ibercom)<br>
* ASTERISK-24362 - res_hep leaks reference to configuration<br>
(Reported by Corey Farrell)<br>
* ASTERISK-23781 - outgoing missing as enum from<br>
contrib/ast-db-manage/config (Reported by Stephen More)<br>
* ASTERISK-24199 - 'ALL' is specified in pjsip.conf.sample for TLS<br>
cipher but it is not valid (Reported by Joshua Colp)<br>
* ASTERISK-24262 - AMI CoreShowChannel missing several output<br>
fields and event documentation (Reported by Mitch Claborn)<br>
* ASTERISK-24356 - PJSIP: Directed pickup causes deadlock<br>
(Reported by Richard Mudgett)<br>
* ASTERISK-24195 - bridge_native_rtp: Removing mixmonitor from a<br>
native RTP capable smart bridge doesn't cause the bridge to<br>
resume being a native rtp bridge (Reported by Jonathan Rose)<br>
* ASTERISK-24384 - chan_motif: format capabilities leak on module<br>
load error (Reported by Corey Farrell)<br>
* ASTERISK-24385 - chan_sip: process_sdp leaks on an error path<br>
(Reported by Corey Farrell)<br>
* ASTERISK-24378 - Release AMI connections on shutdown (Reported<br>
by Corey Farrell)<br>
* ASTERISK-24369 - res_pjsip: Large message on reliable transport<br>
can cause empty messages to be passed from the PJSIP stack up,<br>
causing crashes in multiple locations (Reported by Matt Jordan)<br>
* ASTERISK-24382 - chan_pjsip: Calling PJSIP_MEDIA_OFFER on a<br>
non-PJSIP channel results in an invalid reference of a channel<br>
pvt and a FRACK (Reported by Matt Jordan)<br>
* ASTERISK-24370 - res_pjsip/pjsip_options: OPTIONS request sent<br>
to Asterisk with no user in request is always 404'd (Reported by<br>
Matt Jordan)<br>
* ASTERISK-24224 - When using Bridge() dialplan application,<br>
surrogate channel appears in list and call count is inflated.<br>
(Reported by Mark Michelson)<br>
* ASTERISK-24354 - AMI sendMessage closes AMI connection on error<br>
(Reported by Peter Katzmann)<br>
* ASTERISK-24398 - Initialize auth_rejection_permanent on client<br>
state to the configuration parameter value (Reported by Matt<br>
Jordan)<br>
* ASTERISK-24326 - res_rtp_asterisk: ICE-TCP candidates are<br>
incorrectly attempted (Reported by Joshua Colp)<br>
* ASTERISK-24011 - [patch]safe_asterisk tries to set ulimit -n too<br>
high on linux systems with lots of RAM (Reported by Michael<br>
Myles)<br>
* ASTERISK-24383 - res_rtp_asterisk: Crash if no candidates<br>
received for component (Reported by Kevin Harwell)<br>
* ASTERISK-20784 - Failure to receive an ACK to a SIP Re-INVITE<br>
results in a SIP channel leak (Reported by NITESH BANSAL)<br>
* ASTERISK-15879 - [patch] Failure to receive an ACK to a SIP<br>
Re-INVITE results in a SIP channel leak (Reported by Torrey<br>
Searle)<br>
* ASTERISK-24387 - res_pjsip: rport sent from UAS MUST include the<br>
port that the UAC sent the request on (Reported by Matt Jordan)<br>
* ASTERISK-24406 - Some caller ID strings are parsed differently<br>
since 11.13.0 (Reported by Etienne Lessard)<br>
* ASTERISK-24325 - res_calendar_ews: cannot be used with neon 0.30<br>
(Reported by Tzafrir Cohen)<br>
* ASTERISK-13797 - [patch] relax badshell tilde test (Reported by<br>
Tzafrir Cohen)<br>
* ASTERISK-22791 - asterisk sends Re-INVITE after receiving a BYE<br>
(Reported by Paolo Compagnini)<br>
* ASTERISK-18923 - res_fax_spandsp usage counter is wrong<br>
(Reported by Grigoriy Puzankin)<br>
* ASTERISK-24394 - CDR: FRACK with PJSIP directed pickup.<br>
(Reported by Richard Mudgett)<br>
* ASTERISK-24392 - res_fax: fax gateway sessions leak (Reported by<br>
Corey Farrell)<br>
* ASTERISK-24321 - SIP deadlock when running automated queues<br>
tests (Reported by Steve Pitts)<br>
* ASTERISK-24393 - rtptimeout=0 doesn't disable rtptimeout<br>
(Reported by Dmitry Melekhov)<br>
* ASTERISK-23846 - Unistim multilines. Loss of voice after second<br>
call drops (on a second line). (Reported by Rustam Khankishyiev)<br>
* ASTERISK-24312 - SIGABRT when improperly configured realtime<br>
pjsip (Reported by Dafi Ni)<br>
* ASTERISK-24426 - CDR Batch mode: size used as time value after<br>
first expire (Reported by Shane Blaser)<br>
* ASTERISK-24327 - bridge_native_rtp: Smart bridge operation to<br>
softmix sometimes fails to properly re-INVITE remotely bridged<br>
participants (Reported by Matt Jordan)<br>
* ASTERISK-24415 - Missing AMI VarSet events when channels inherit<br>
variables. (Reported by Richard Mudgett)<br>
* ASTERISK-24063 - [patch]Asterisk does not respect outbound proxy<br>
when sending qualify requests (Reported by Damian Ivereigh)<br>
* ASTERISK-24122 - Documentaton for res_pjsip option use_avpf<br>
needs to be fixed (Reported by James Van Vleet)<br>
* ASTERISK-24381 - res_pjsip_sdp_rtp: Declined media streams are<br>
interpreted, leading to erroneous 488 rejections (Reported by<br>
Matt Jordan)<br>
* ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of<br>
SSLv3, security fix POODLE (CVE-2014-3566) (Reported by<br>
abelbeck)<br>
* ASTERISK-24436 - Missing header in res/res_srtp.c when compiling<br>
against libsrtp-1.5.0 (Reported by Patrick Laimbock)<br>
* ASTERISK-24454 - app_queue: ao2_iterator not destroyed, causing<br>
leak (Reported by Corey Farrell)<br>
* ASTERISK-24430 - missing letter "p" in word response in<br>
OriginateResponse event documentation (Reported by Dafi Ni)<br>
* ASTERISK-24437 - Review implementation of ast_bridge_impart for<br>
leaks and document proper usage (Reported by Scott Griepentrog)<br>
* ASTERISK-24453 - manager: acl_change_sub leaks (Reported by<br>
Corey Farrell)<br>
* ASTERISK-24457 - res_fax: fax gateway frames leak (Reported by<br>
Corey Farrell)<br>
* ASTERISK-21721 - SIP Failed to parse multiple Supported: headers<br>
(Reported by Olle Johansson)<br>
* ASTERISK-24304 - asterisk crashing randomly because of unistim<br>
channel (Reported by dhanapathy sathya)<br>
* ASTERISK-24190 - IMAP voicemail causes segfault (Reported by<br>
Nick Adams)<br>
* ASTERISK-24462 - res_pjsip: Stale qualify statistics after<br>
disablementation (Reported by Kevin Harwell)<br>
* ASTERISK-24466 - app_queue: fix a couple leaks to struct<br>
call_queue (Reported by Corey Farrell)<br>
* ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled<br>
(Reported by Corey Farrell)<br>
* ASTERISK-24411 - [patch] Status of outbound registration is not<br>
changed upon unregistering. (Reported by John Bigelow)<br>
* ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream<br>
leaks (Reported by Corey Farrell)<br>
* ASTERISK-24487 - configuration: sections should be loadable as<br>
template even when not marked (Reported by Scott Griepentrog)<br>
* ASTERISK-24307 - Unintentional memory retention in stringfields<br>
(Reported by Etienne Lessard)<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-12.7.0" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.7.0</a><br>
<br>
Thank you for your continued support of Asterisk!<br>
<br>
<br>--<br>
_____________________________________________________________________<br>
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