<div dir="ltr"><div class="gmail_extra"><div><div dir="ltr"><div>Jorge,</div><div><br></div><div>This is a function of Asterisk, not Opensips. This page may help you:</div><div> <a href="http://www.voztovoice.org/?q=node/350">http://www.voztovoice.org/?q=node/350</a></div>
<div><br></div><div><br></div><div>- Jeff</div><div><br></div></div></div>
<br><br><div class="gmail_quote">On Mon, Dec 16, 2013 at 7:00 PM, Jorge Ortea <span dir="ltr"><<a href="mailto:darham@hotmail.com" target="_blank">darham@hotmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<div><div dir="ltr">Hi all,<div><br></div><div>Suppose a platform with OpenSIPS and several Asterisk behind. A new call in a Asterisk that send to Opensips to route to uac1. The uac1 is ringing, it is sending 180 Ringing, then from other uac wants CallPickup this call, this feature is dialed but when the Invite reach to OpenSIPS,,, How I can know that Asterisk is the call?</div>
<div><br></div><div><br></div><div>Very Thanks.</div><div>Regards.</div> </div></div>
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