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: 2012/11/5<br>Subject: [asterisk-dev] Asterisk 11.0.1 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 release of Asterisk 11.0.1.<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 11.0.1 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>
* --- chan_sip: Fix a bug causing SIP reloads to remove all entries<br>
from the registry<br>
(Closes issue ASTERISK-20611. Reported by Alisher)<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 an issue with res_http_websocket where the chan_sip<br>
WebSocket handler could not be registered.<br>
(Closes issue ASTERISK-20631. Reported by danjenkins)<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-11.0.1" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.1</a><br>
<br>
Thank you for your continued support of Asterisk!<br>
<br>
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