<div dir="ltr">Sorry if this is a stupid question:<div><br></div><div>I've connected to FreeSwicth using mod_verto, now I would like to route the call to a Genesys via SIP. </div><div><br></div><div>Is it the right approach? is there any documentation for it? </div><div><br></div><div>The purpose is to be able to integrate my webRTC solution to many contact center platforms.<br><div class="gmail_signature"><div style="width:400px;float:left"><table cellpadding="0" cellspacing="0" border="0"><tbody><tr><td valign="top" colspan="2"><br></td></tr><tr><td valign="top" colspan="2" style="padding-top:19px"><br></td></tr><tr><td valign="top" colspan="2"><br></td></tr><tr><td valign="top" style="padding-top:10px"><br></td></tr><tr><td valign="top" colspan="2" style="padding-top:20px"><br></td></tr><tr><td valign="top" colspan="2" style="padding-top:20px"><br></td></tr></tbody></table>
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