<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 14:06<br>Assunto: [asterisk-dev] Asterisk 12.2.0 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 12.2.0.<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 12.2.0 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-23276 - Look at adding the 'pjsip show channel' command<br>
(Reported by George Joseph)<br>
<br>
Bugs fixed in this release:<br>
-----------------------------------<br>
* ASTERISK-23290 - chan_sip: ast_bridge_transfer_blind causes<br>
channel to be hung up immediately, leading to BYE request being<br>
sent before NOTIFY (Reported by Matt Jordan)<br>
* ASTERISK-23098 - [patch]possible null pointer dereference in<br>
format.c (Reported by Marcello Ceschia)<br>
* ASTERISK-23125 - ARI: URI is case sensitive (Reported by Zane<br>
Conkle)<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-22738 - "Security denial" error in calls from H323<br>
trunk (ooh323.c) (Reported by Gabriele Odone)<br>
* ASTERISK-23069 - Custom CDR variable not recorded when set in<br>
macro called from app_queue (Reported by Bryan Anderson)<br>
* ASTERISK-23266 - [patch]pjsip_cli: Memory leak in<br>
ast_sip_cli_print_sorcery_objectset (Reported by George Joseph)<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-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-23320 - Preloading pbx_config.so with a CustomPresence<br>
hint defined results in crash (Reported by xrobau)<br>
* ASTERISK-23265 - Preloading Certain Modules in Asterisk 12<br>
causes a core dump (Reported by Andrew Nagy)<br>
* ASTERISK-23287 - res_pjsip_refer: Crash during attended transfer<br>
when attended->transferer_second channel is NULL (Reported by<br>
Matt Jordan)<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-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-23258 - Target_uri for LiveRecording model in ARI<br>
(Reported by Ben Merrills)<br>
* ASTERISK-17523 - Qualify for static realtime peers does not work<br>
(Reported by Maciej Krajewski)<br>
* ASTERISK-23204 - Device state cache requires improvement<br>
(Reported by Mark Michelson)<br>
* ASTERISK-23092 - cli: pjsip show endpoint <endpoint> shows<br>
allow/disallow codecs the same (Reported by Dan Jenkins)<br>
* ASTERISK-21406 - [patch] chan_sip deadlock on monlock between<br>
unload_module and do_monitor (Reported by Corey Farrell)<br>
* ASTERISK-23210 - Security: Remote crash in res_pjsip. (Reported<br>
by Joshua Colp)<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-23020 - PJSip - Multihomed machine returning wrong IP<br>
address (Reported by xrobau)<br>
* ASTERISK-23311 - Manager - MoH Stop Event fails to show up when<br>
leaving Conference (Reported by Benjamin Keith Ford)<br>
* ASTERISK-23295 - ARI: ChannelEnteredBridge event not delivered<br>
to client during bridge move operation (Reported by Matt Jordan)<br>
* ASTERISK-23444 - Playback and Record events not subscribed to<br>
when calling Play/Record on bridge (Reported by Ben Merrills)<br>
* ASTERISK-23235 - pjsip transport/tos interpreted differently<br>
than endpoint/tos_audio (Reported by George Joseph)<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-23254 - Bad ao2_find() usage in pjsip_options.c<br>
(Reported by Richard Mudgett)<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-22537 - Create Sorcery equivalent to the AST_CONFIG<br>
function (Reported by George Joseph)<br>
* ASTERISK-23275 - CLI command 'pjsip show registrations' missing<br>
(Reported by George Joseph)<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>
* ASTERISK-23233 - alembic missing scripts for certain realtime<br>
tables (Reported by jmls)<br>
* ASTERISK-22537 - Create Sorcery equivalent to the AST_CONFIG<br>
function (Reported by George Joseph)<br>
* ASTERISK-23120 - ARI/AMI: allow objects created via interfaces<br>
to have their unique ID specified by the external application<br>
(Reported by Matt Jordan)<br>
* ASTERISK-22008 - Config framework docs should display the<br>
see-also information in CLI output. (Reported by Richard<br>
Mudgett)<br>
* ASTERISK-23435 - PJSIP: Fix the DNS resolution (whoops)<br>
(Reported by Matt Jordan)<br>
* ASTERISK-22499 - ARI documentation - point to HTTP server<br>
configuration sample and wiki docs where appropriate (Reported<br>
by Rusty Newton)<br>
* ASTERISK-23437 - ARI: Add the ability to add properties to a<br>
bridge on creation (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-12.2.0" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-12.2.0</a><br>
<br>
Thank you for your continued support of Asterisk!<br>
<br>
<br>--<br>
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