<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_11990" dir="ltr">that you have written is like me, or do I mistake?</div><div id="yui_3_16_0_1_1424849201856_11990" dir="ltr"><span><br></span></div><div id="yui_3_16_0_1_1424849201856_11990" dir="ltr"><span>thanks</span></div><div></div><div id="yui_3_16_0_1_1424849201856_12007"> </div><div class="signature" id="yui_3_16_0_1_1424849201856_12005"><div id="yui_3_16_0_1_1424849201856_12004"><span style="font-family:arial, helvetica, sans-serif;color:rgb(208, 208, 208);" id="yui_3_16_0_1_1424849201856_12006">_______________________________________________________________________</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);" id="yui_3_16_0_1_1424849201856_12078">Oscar OXY</span><br></div></div><br> <div style="font-family: verdana, helvetica, sans-serif; font-size: 13px;" id="yui_3_16_0_1_1424849201856_11994"> <div style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 16px;" id="yui_3_16_0_1_1424849201856_11993"> <div dir="ltr" id="yui_3_16_0_1_1424849201856_11992"> <hr size="1" id="yui_3_16_0_1_1424849201856_12079"> <font size="2" face="Arial" id="yui_3_16_0_1_1424849201856_11991"> <b><span style="font-weight:bold;">Da:</span></b> Ben Merrills <b.merrills@mersontech.co.uk><br> <b><span style="font-weight: bold;">A:</span></b> Asterisk Application Development discussion <asterisk-app-dev@lists.digium.com> <br> <b><span style="font-weight: bold;">Inviato:</span></b> Mercoledì 25 Febbraio 2015 9:50<br> <b><span style="font-weight: bold;">Oggetto:</span></b> Re: [asterisk-app-dev] How do I bind a outbound call?<br> </font> </div> <div class="y_msg_container" id="yui_3_16_0_1_1424849201856_11995"><br>> I'm doing a predictive dialer by ARI, that is:<br clear="none">><br clear="none">> asterisk call 2 number phone (to PSTN) in outbound, when one phone PSTN answer (we suppose answer first phone) then asterisk call my sip phone, when I answer then asterisk bind, by a bridge, the first channel (PSTN) and my sip phone and hang up the second calling (PSTN).<br clear="none">><br clear="none">> is it right? exist a ARI function redirect from PSTN to my sip phone without bridge?<br clear="none">> thanks<br clear="none"><br clear="none">The general pattern here would be, in my opinion.<br clear="none"><br clear="none">Originate 1st channel (PSTN)<br clear="none">On Answer Originate 2nd channel (local sip extension)<br clear="none">On Answer Create a bridge and add 1st and second channels to the bridge<br clear="none">On Hang-up destroy bridge<div class="qtdSeparateBR"><br><br></div><div class="yqt6458326939" id="yqtfd69828"><br clear="none"><br clear="none"> <br clear="none">_______________________________________________________________________<br clear="none">Oscar OXY<br clear="none">________________________________________<br clear="none">_______________________________________________<br clear="none">asterisk-app-dev mailing list<br clear="none"><a shape="rect" ymailto="mailto:asterisk-app-dev@lists.digium.com" href="mailto:asterisk-app-dev@lists.digium.com">asterisk-app-dev@lists.digium.com</a><br clear="none"><a shape="rect" href="http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev" target="_blank">http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev</a><br clear="none"></div><br><br></div> </div> </div> </div></body></html>