<html><body><div style="color:#000; background-color:#fff; font-family:verdana, helvetica, sans-serif;font-size:13px"><div id="yui_3_16_0_1_1424849201856_3834"><span></span></div><div id="yui_3_16_0_1_1424849201856_3834" dir="ltr"><span id="yui_3_16_0_1_1424849201856_3918">I'm doing a predictive dialer by ARI, that is:</span></div><div id="yui_3_16_0_1_1424849201856_3834" dir="ltr"><span><br></span></div><div id="yui_3_16_0_1_1424849201856_3834" dir="ltr"><span id="yui_3_16_0_1_1424849201856_3927" class="" style="">asterisk call 2 number phone (to PSTN) in outbound, when one phone PSTN </span>answer <span class="" style="" id="yui_3_16_0_1_1424849201856_4115">(we suppose answer first phone) then asterisk call my sip phone, when I answer then asterisk bind, by a bridge, the first channel (</span>PSTN) and my sip phone and hang up the second calling (PSTN).</div><div id="yui_3_16_0_1_1424849201856_3834" dir="ltr"><span><br></span></div><div id="yui_3_16_0_1_1424849201856_3834" dir="ltr">is it right? exist a ARI function redirect from PSTN to my sip phone without bridge?</div><div id="yui_3_16_0_1_1424849201856_3834" dir="ltr">thanks</div><div id="yui_3_16_0_1_1424849201856_3834" dir="ltr"><span><br></span></div><div id="yui_3_16_0_1_1424849201856_3834" dir="ltr"><span><br></span></div><div></div><div id="yui_3_16_0_1_1424849201856_3864"> </div><div class="signature" id="yui_3_16_0_1_1424849201856_3839"><div id="yui_3_16_0_1_1424849201856_3838"><span style="font-family:arial, helvetica, sans-serif;color:rgb(208, 208, 208);" id="yui_3_16_0_1_1424849201856_3837">_______________________________________________________________________</span><br style="font-family:arial, helvetica, sans-serif;color:rgb(208, 208, 208);"><span style="font-family:arial, helvetica, sans-serif;color:rgb(208, 208, 208);">Oscar OXY</span><br></div></div></div></body></html>