<div class="gmail_quote">---------- Mensagem encaminhada ----------<br>De: "Asterisk Development Team" <<a href="mailto:asteriskteam@digium.com">asteriskteam@digium.com</a>><br>Data: 23/04/2014 14:05<br>Assunto: [asterisk-dev] Asterisk 11.9.0 Now Available<br>
Para: "Asterisk Developers Mailing List" <<a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br>Cc: <br><br type="attribution">The Asterisk Development Team has announced the release of Asterisk 11.9.0.<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 11.9.0 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-22790 - check_modem_rate() may return incorrect rate<br>
for V.27 (Reported by Paolo Compagnini)<br>
* ASTERISK-23034 - [patch] manager Originate doesn't abort on<br>
failed format_cap allocation (Reported by Corey Farrell)<br>
* ASTERISK-23061 - [Patch] 'textsupport' setting not mentioned in<br>
sip.conf.sample (Reported by Eugene)<br>
* ASTERISK-23028 - [patch] Asterisk man pages contains unquoted<br>
minus signs (Reported by Jeremy Lainé)<br>
* ASTERISK-23046 - Custom CDR fields set during a GoSUB called<br>
from app_queue are not inserted (Reported by Denis Pantsyrev)<br>
* ASTERISK-23027 - [patch] Spelling typo "transfered" instead of<br>
"transferred" (Reported by Jeremy Lainé)<br>
* ASTERISK-23008 - Local channels loose CALLERID name when DAHDI<br>
channel connects (Reported by Michael Cargile)<br>
* ASTERISK-23100 - [patch] In chan_mgcp the ident in transmitted<br>
request and request queue may differ - fix for locking (Reported<br>
by adomjan)<br>
* ASTERISK-22988 - [patch]T38 , SIP 488 after Rejecting image<br>
media offer due to invalid or unsupported syntax (Reported by<br>
adomjan)<br>
* ASTERISK-22861 - [patch]Specifying a null time as parameter to<br>
GotoIfTime or ExecIfTime causes segmentation fault (Reported by<br>
Sebastian Murray-Roberts)<br>
* ASTERISK-17837 - extconfig.conf - Maximum Include level (1)<br>
exceeded (Reported by pz)<br>
* ASTERISK-22662 - Documentation fix? - queues.conf says<br>
persistentmembers defaults to yes, it appears to lie (Reported<br>
by Rusty Newton)<br>
* ASTERISK-23134 - [patch] res_rtp_asterisk port selection cannot<br>
handle selinux port restrictions (Reported by Corey Farrell)<br>
* ASTERISK-23220 - STACK_PEEK function with no arguments causes<br>
crash/core dump (Reported by James Sharp)<br>
* ASTERISK-19773 - Asterisk crash on issuing Asterisk-CLI 'reload'<br>
command multiple times on cli_aliases (Reported by Joel Vandal)<br>
* ASTERISK-22757 - segfault in res_clialiases.so on reload when<br>
mapping "module reload" command (Reported by Gareth Blades)<br>
* ASTERISK-17727 - [patch] TLS doesn't get all certificate chain<br>
(Reported by LN)<br>
* ASTERISK-23178 - devicestate.h: device state setting functions<br>
are documented with the wrong return values (Reported by<br>
Jonathan Rose)<br>
* ASTERISK-23232 - LocalBridge AMI Event LocalOptimization value<br>
is opposite to what's expected (Reported by Leon Roy)<br>
* ASTERISK-23098 - [patch]possible null pointer dereference in<br>
format.c (Reported by Marcello Ceschia)<br>
* ASTERISK-23297 - Asterisk 12, pbx_config.so segfaults if<br>
res_parking.so is not loaded, or if res_parking.conf has no<br>
configuration (Reported by CJ Oster)<br>
* ASTERISK-23069 - Custom CDR variable not recorded when set in<br>
macro called from app_queue (Reported by Bryan Anderson)<br>
* ASTERISK-19499 - ConfBridge MOH is not working for transferee<br>
after attended transfer (Reported by Timo Teräs)<br>
* ASTERISK-23261 - [patch]Output mixup in<br>
${CHANNEL(rtpqos,audio,all)} (Reported by rsw686)<br>
* ASTERISK-23279 - [patch]Asterisk doesn't support the dynamic<br>
payload change in rtp mapping in the 200 OK response (Reported<br>
by NITESH BANSAL)<br>
* ASTERISK-23255 - UUID included for Redhat, but missing for<br>
Debian distros in install_prereq script (Reported by Rusty<br>
Newton)<br>
* ASTERISK-23260 - [patch]ForkCDR v option does not keep CDR<br>
variables for subsequent records (Reported by zvision)<br>
* ASTERISK-23141 - Asterisk crashes on Dial(), in<br>
pbx_find_extension at pbx.c (Reported by Maxim)<br>
* ASTERISK-23336 - Asterisk warning "Don't know how to indicate<br>
condition 33 on ooh323c" on outgoing calls from H323 to SIP peer<br>
(Reported by Alexander Semych)<br>
* ASTERISK-23231 - Since 405693 If we have res_fax.conf file set<br>
to minrate=2400, then res_fax refuse to load (Reported by David<br>
Brillert)<br>
* ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set<br>
- probably introduced in 11.7.0 (Reported by OK)<br>
* ASTERISK-23323 - [patch]chan_sip: missing p->owner checks in<br>
handle_response_invite (Reported by Walter Doekes)<br>
* ASTERISK-23406 - [patch]Fix typo in "sip show peer" (Reported by<br>
ibercom)<br>
* ASTERISK-23310 - bridged channel crashes in bridge_p2p_rtp_write<br>
(Reported by Jeremy Lainé)<br>
* ASTERISK-22911 - [patch]Asterisk fails to resume WebRTC call<br>
from hold (Reported by Vytis Valentinavičius)<br>
* ASTERISK-23104 - Specifying the SetVar AMI without a Channel<br>
cause Asterisk to crash (Reported by Joel Vandal)<br>
* ASTERISK-21930 - [patch]WebRTC over WSS is not working.<br>
(Reported by John)<br>
* ASTERISK-23383 - Wrong sense test on stat return code causes<br>
unchanged config check to break with include files. (Reported by<br>
David Woolley)<br>
* ASTERISK-20149 - Crash when faxing SIP to SIP with strictrtp set<br>
to yes (Reported by Alexandr Gordeev)<br>
* ASTERISK-17523 - Qualify for static realtime peers does not work<br>
(Reported by Maciej Krajewski)<br>
* ASTERISK-21406 - [patch] chan_sip deadlock on monlock between<br>
unload_module and do_monitor (Reported by Corey Farrell)<br>
* ASTERISK-23373 - [patch]Security: Open FD exhaustion with<br>
chan_sip Session-Timers (Reported by Corey Farrell)<br>
* ASTERISK-23340 - Security Vulnerability: stack allocation of<br>
cookie headers in loop allows for unauthenticated remote denial<br>
of service attack (Reported by Matt Jordan)<br>
* ASTERISK-23311 - Manager - MoH Stop Event fails to show up when<br>
leaving Conference (Reported by Benjamin Keith Ford)<br>
* ASTERISK-23420 - [patch]Memory leak in manager_add_filter<br>
function in manager.c (Reported by Etienne Lessard)<br>
* ASTERISK-23488 - Logic error in callerid checksum processing<br>
(Reported by Russ Meyerriecks)<br>
* ASTERISK-23461 - Only first user is muted when joining<br>
confbridge with 'startmuted=yes' (Reported by Chico Manobela)<br>
* ASTERISK-20841 - fromdomain not honored on outbound INVITE<br>
request (Reported by Kelly Goedert)<br>
* ASTERISK-22079 - Segfault: INTERNAL_OBJ (user_data=0x6374652f)<br>
at astobj2.c:120 (Reported by Jamuel Starkey)<br>
* ASTERISK-23509 - [patch]SayNumber for Polish language tries to<br>
play empty files for numbers divisible by 100 (Reported by<br>
zvision)<br>
* ASTERISK-23103 - [patch]Crash in ast_format_cmp, in ao2_find<br>
(Reported by JoshE)<br>
* ASTERISK-23391 - Audit dialplan function usage of channel<br>
variable (Reported by Corey Farrell)<br>
* ASTERISK-23548 - POST to ARI sometimes returns no body on<br>
success (Reported by Scott Griepentrog)<br>
* ASTERISK-23460 - ooh323 channel stuck if call is placed directly<br>
and gatekeeper is not available (Reported by Dmitry Melekhov)<br>
<br>
Improvements made in this release:<br>
-----------------------------------<br>
* ASTERISK-22980 - [patch]Allow building cdr_radius and cel_radius<br>
against libfreeradius-client (Reported by Jeremy Lainé)<br>
* ASTERISK-22661 - Unable to exit ChanSpy if spied channel does<br>
not have a call in progress (Reported by Chris Hillman)<br>
* ASTERISK-23099 - [patch] WSS: enable ast_websocket_read()<br>
function to read the whole available data at first and then wait<br>
for any fragmented packets (Reported by Thava Iyer)<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-11.9.0" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.9.0</a><br>
<br>
Thank you for your continued support of Asterisk!<br>
<br>
<br>--<br>
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