<br><div class="gmail_quote">Hi All,<br><br>We have a Tandberg VCS System for Video conferencing and a customer running AsteriskNow (Asterisk 1.6 + FreePBX) for Audio conferencing.<br><br>Problem Statement:<br>How do we integrate the 2 systems such that Audio SIP calls are seamlessly passed between the two. Sorry we're just starting up so a bit of general advice, or a link to any document would be great!<br>
<br>If anybody has done this - would appreciate any tips :)<br><br><br>Thanks!<br><font color="#888888"><br><br>Jake<br><br>
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