<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;">24 de agosto de 2010 12:35:14 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;"><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 1.8.0-beta4 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 1.8.0-beta4.<br>This 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>All interested users of Asterisk are encouraged to participate in the 1.8<br>testing process. Please report any issues found to the issue tracker,<br>http://issues.asterisk.org/. It is also very useful to see successful test<br>reports. Please post those to the asterisk-dev mailing list.<br><br>Asterisk 1.8 is the next major release series of Asterisk. It will be a Long<br>Term Support (LTS) release, similar to Asterisk 1.4. For more information about<br>support time lines for Asterisk releases, see the Asterisk versions page.<br><br>http://www.asterisk.org/asterisk-versions<br><br>This release contains fixes since the last beta release as reported by the<br>community. A sampling of the changes in this release include:<br><br> * Fix parsing of IPv6 address literals in outboundproxy<br> (Closes issue #17757. Reported by oej. Patched by sperreault)<br><br> * Change the default value for alwaysauthreject in sip.conf to "yes".<br> (Closes issue #17756. Reported by oej)<br><br> * Remove current STUN support from chan_sip.c. This change removes the current<br> broken/useless STUN support from chan_sip.<br> (Closes issue #17622. Reported by philipp2.<br> Review: https://reviewboard.asterisk.org/r/855/)<br><br> * PRI CCSS may use a stale dial string for the recall dial string. If an<br> outgoing call negotiates a different B channel than initially requested, the<br> saved original dial string was not transferred to the new B channel. CCSS<br> uses that dial string to generate the recall dial string.<br> (Patched by rmudgett)<br><br> * Split _all_ arguments before parsing them. This fixes multicast RTP paging<br> using linksys mode.<br> (Patched by russellb)<br><br> * Expand cel_custom.conf.sample. Include the usage of CSV_QUOTE() to ensure<br> data has valid CSV formatting. Also list the special CEL variables that are<br> available for use in the mapping. There are also several other CEL fixes in<br> this release.<br> (Patched by russellb)<br><br><br>Asterisk 1.8 contains many new features over previous releases of Asterisk.<br>A short list of included features includes:<br><br> * Secure RTP<br> * IPv6 Support in the SIP Channel<br> * Connected Party Identification Support<br> * Calendaring Integration<br> * A new call logging system, Channel Event Logging (CEL)<br> * Distributed Device State using Jabber/XMPP PubSub<br> * Call Completion Supplementary Services support<br> * Advice of Charge support<br> * Much, much more!<br><br>A full list of new features can be found in the CHANGES file.<br><br>http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout<br><br>For a full list of changes in the current release, please see the ChangeLog:<br><br>http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta4<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>