<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><br><div><br><div>Begin forwarded message:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, 0, 1);"><b>From: </b></span><span style="font-family:'Helvetica'; font-size:medium;">Asterisk Development Team <<a href="mailto:asteriskteam@digium.com">asteriskteam@digium.com</a>><br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, 0, 1);"><b>Date: </b></span><span style="font-family:'Helvetica'; font-size:medium;">10 de maio de 2011 11:38:48 BRT<br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, 0, 1);"><b>To: </b></span><span style="font-family:'Helvetica'; font-size:medium;">Asterisk Development Team <<a href="mailto:asteriskteam@digium.com">asteriskteam@digium.com</a>><br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, 0, 1);"><b>Subject: </b></span><span style="font-family:'Helvetica'; font-size:medium;"><b>[asterisk-dev] Asterisk 1.8.4 Now Available</b><br></span></div><div style="margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style="font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, 0, 1);"><b>Reply-To: </b></span><span style="font-family:'Helvetica'; font-size:medium;">Asterisk Developers Mailing List <<a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br></span></div><br><div>The Asterisk Development Team has announced the release of Asterisk 1.8.4. This<br>release is available for immediate download at<br><a href="http://downloads.asterisk.org/pub/telephony/asterisk/">http://downloads.asterisk.org/pub/telephony/asterisk/</a><br><br>The release of Asterisk 1.8.4 resolves several issues reported by the community.<br>Without your help this release would not have been possible. Thank you!<br><br>Below is a sample of the issues resolved in this release:<br><br> * Use SSLv23_client_method instead of old SSLv2 only.<br> (Closes issue #19095, #19138. Reported, patched by tzafrir. Tested by russell<br> and chazzam.<br><br> * Resolve crash in ast_mutex_init()<br> (Patched by twilson)<br><br> * Resolution of several DTMF based attended transfer issues.<br> (Closes issue #17999, #17096, #18395, #17273. Reported by iskatel, gelo,<br> shihchuan, grecco. Patched by rmudgett)<br><br> NOTE: Be sure to read the ChangeLog for more information about these changes.<br><br> * Resolve deadlocks related to device states in chan_sip<br> (Closes issue #18310. Reported, patched by one47. Patched by jpeeler)<br><br> * Resolve an issue with the Asterisk manager interface leaking memory when<br> disabled.<br> (Reported internally by kmorgan. Patched by russellb)<br><br> * Support greetingsfolder as documented in voicemail.conf.sample.<br> (Closes issue #17870. Reported by edhorton. Patched by seanbright)<br><br> * Fix channel redirect out of MeetMe() and other issues with channel softhangup<br> (Closes issue #18585. Reported by oej. Tested by oej, wedhorn, russellb.<br> Patched by russellb)<br><br> * Fix voicemail sequencing for file based storage.<br> (Closes issue #18498, #18486. Reported by JJCinAZ, bluefox. Patched by<br> jpeeler)<br><br> * Set hangup cause in local_hangup so the proper return code of 486 instead of<br> 503 when using Local channels when the far sides returns a busy. Also affects<br> CCSS in Asterisk 1.8+.<br> (Patched by twilson)<br><br> * Fix issues with verbose messages not being output to the console.<br> (Closes issue #18580. Reported by pabelanger. Patched by qwell)<br><br> * Fix Deadlock with attended transfer of SIP call<br> (Closes issue #18837. Reported, patched by alecdavis. Tested by<br> alecdavid, Irontec, ZX81, cmaj)<br><br>Includes changes per AST-2011-005 and AST-2011-006<br>For a full list of changes in this release candidate, please see the ChangeLog:<br><br><a href="http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4">http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4</a><br><br>Information about the security releases are available at:<br><br>http://downloads.asterisk.org/pub/security/AST-2011-005.pdf<br>http://downloads.asterisk.org/pub/security/AST-2011-006.pdf<br><br>Thank you for your continued support of Asterisk!<br><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by http://www.api-digital.com --<br><br>asterisk-dev mailing list<br>To UNSUBSCRIBE or update options visit:<br> http://lists.digium.com/mailman/listinfo/asterisk-dev<br></div></blockquote></div><br></body></html>