<html><body><div style="color:#000; background-color:#fff; font-family:verdana, helvetica, sans-serif;font-size:13px"><div dir="ltr" id="yui_3_16_0_1_1424849201856_22419"><span id="yui_3_16_0_1_1424849201856_22418">I'm sorry for my bed english ...</span></div><div dir="ltr" id="yui_3_16_0_1_1424849201856_22425"><span id="yui_3_16_0_1_1424849201856_22424">but in wiki there is:</span></div><div dir="ltr" id="yui_3_16_0_1_1424849201856_22417"><span><br></span></div><div dir="ltr" id="yui_3_16_0_1_1424849201856_22386"><span id="yui_3_16_0_1_1424849201856_22385"><a href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Channels+REST+API#Asterisk13ChannelsRESTAPI-redirect" id="yui_3_16_0_1_1424849201856_22384">https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Channels+REST+API#Asterisk13ChannelsRESTAPI-redirect</a><br></span></div><div dir="ltr" id="yui_3_16_0_1_1424849201856_22386"><br></div><div dir="ltr" id="yui_3_16_0_1_1424849201856_22386">so I thought that I could do a originate call by ARI and deviate it from PSTN to sip without bridge, but only when originate call answer.</div><div dir="ltr" id="yui_3_16_0_1_1424849201856_22386">is it more clear?</div><div dir="ltr" id="yui_3_16_0_1_1424849201856_22386"><br></div><div dir="ltr" id="yui_3_16_0_1_1424849201856_22386">ps. I wouldn't use AMI, but only ARI so I manage anything by WebSocket.</div><div dir="ltr" id="yui_3_16_0_1_1424849201856_22386"><br></div><div dir="ltr" id="yui_3_16_0_1_1424849201856_22386">thanks</div><div></div><div id="yui_3_16_0_1_1424849201856_22426"> </div><div class="signature" id="yui_3_16_0_1_1424849201856_22373"><div id="yui_3_16_0_1_1424849201856_22372"><span style="font-family:arial, helvetica, sans-serif;color:rgb(208, 208, 208);" id="yui_3_16_0_1_1424849201856_22379">_______________________________________________________________________</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><br> <div style="font-family: verdana, helvetica, sans-serif; font-size: 13px;" id="yui_3_16_0_1_1424849201856_22376"> <div style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 16px;" id="yui_3_16_0_1_1424849201856_22375"> <div dir="ltr" id="yui_3_16_0_1_1424849201856_22374"> <hr size="1" id="yui_3_16_0_1_1424849201856_22378"> <font size="2" face="Arial" id="yui_3_16_0_1_1424849201856_22377"> <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 10:23<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_22532"><br>> that you have written is like me, or do I mistake?<br clear="none"><br clear="none">> thanks<br clear="none"><br clear="none">You DO need a bridge, if that's what you're asking. The bridge is used to connect the two channels. ARI Does not support Originate to anything else but a Stasis Application. Unlike AMI's Originate which takes a source and destination (channel/extension/app etc), Stasis will only allow you to originate a channel into a Stasis app. I think the reasoning here was mostly to do with security etc, but either way, you need to bridge both the calls.<br clear="none"><br clear="none">You might just be better off using AMI's originate! It would be easier for you, you could set the PSTN as the first leg and your sip extension as the second.<br clear="none"><br clear="none"><a shape="rect" href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerAction_Originate" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerAction_Originate</a><div class="qtdSeparateBR"><br><br></div><div class="yqt3680137316" id="yqtfd10674"><br clear="none"><br clear="none"><br clear="none">> > 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<br clear="none"><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>