psc<br><br><div class="gmail_quote">---------- Forwarded message ----------<br>From: <b class="gmail_sendername">Asterisk Development Team</b> <span dir="ltr"><<a href="mailto:asteriskteam@digium.com">asteriskteam@digium.com</a>></span><br>
Date: 2012/11/6<br>Subject: [asterisk-dev] Asterisk 10.11.0-rc1 Now Available<br>To: Asterisk Developers Mailing List <<a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br><br><br>The Asterisk Development Team has announced the first release candidate of<br>
Asterisk 10.11.0. This release candidate is available for immediate<br>
download at <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 10.11.0-rc1 resolves several issues reported by the<br>
community and would have not been possible without your participation.<br>
Thank you!<br>
<br>
The following is a sample of the issues resolved in this release candidate:<br>
<br>
* --- Prevent resetting of NATted realtime peer address on reload.<br>
(Closes issue ASTERISK-18203. Reported by daren ferreira)<br>
<br>
* --- Do not use a FILE handle when doing SIP TCP reads.<br>
(Closes issue ASTERISK-20212. Reported by Phil Ciccone)<br>
<br>
* --- Fix ConfBridge crash if no timing module loaded.<br>
(Closes issue ASTERISK-19448. Reported by feyfre)<br>
<br>
* --- confbridge: Fix a bug which made conferences not record with<br>
AMI/CLI commands<br>
(Closes issue ASTERISK-20601. Reported by Vilius)<br>
<br>
* --- Fix execution of 'i' extension due to uninitialized variable.<br>
(Closes issue ASTERISK-20455. Reported by Richard Miller)<br>
<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-10.11.0-rc1" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.11.0-rc1</a><br>
<br>
Thank you for your continued support of Asterisk!<br>
<br>
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