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<body class='hmmessage'><div dir='ltr'><div><br></div><div><div>Duane, you understood perfectly. This is exactly that I need.</div><div><br></div><div>I had already thought in dialog module, I manage transfers with this method. But I thought that this works only when the dialog is established and no before.</div><div><br></div><div>I'll try it and tell you.</div><div><br></div><div>Very thanks.</div><div>Regards.</div></div><br><br><div><hr id="stopSpelling">Date: Mon, 16 Dec 2013 21:26:19 -0600<br>From: duane.larson@gmail.com<br>To: users@lists.opensips.org<br>Subject: Re: [OpenSIPS-Users] Call Pickup Feature<br><br><div dir="ltr">I think there might be a little confusion here and a language barrier. áIf I am understanding Jorge correctly I think he is stating that OpenSIPS is load balancing between many Asterisk servers. áOpenSIPS can do the Call Pickup feature as can Asterisk. áI think the issue is that if OpenSIPS is doing Load balancing and UAC1 is ringing and UAC2 wants to "Call Pickup" that call then OpenSIPS needs to route UAC2's request to the same Asterisk server that UAC1 was called on.<div>
<br></div><div>Hope I understood Jorge correctly. áIf I did then you would need to work with "get_dialog_info" function from the Dialog Module (<a href="http://www.opensips.org/html/docs/modules/1.10.x/dialog.html#id295324" target="_blank">http://www.opensips.org/html/docs/modules/1.10.x/dialog.html#id295324</a>)</div>
<div><br></div><div>So you would need to do something like this</div><div><br></div><div>
á á á ááif ( get_dialog_info("host","$var(x)","caller","$fU") || get_dialog_info("host","$var(x)","caller","$rU") ) {<BR>
á á á á á á á á #route to host $var(x)<BR>
á á á á á á á á $du = "sip:" + $rU + "@" + $var(x) + ":5060";<BR>
á á á á á á á á if ( !match_dialog() )<BR>
á á á á á á á á {<BR>
á á á á á á á á á á á á create_dialog();<BR>
á á á á á á á á }<BR>
á á á á á á á á $dlg_val(host) = $du;<BR>
á á á á }áelse if ( get_dialog_info("host","$var(x)","caller","$rU") || get_dialog_info("host","$var(x)","caller","$fU") ) {<BR>
á á á á á á á áá#route to host $var(x)<BR>á á á á á á á áá$du = "sip:" + $rU + "@" + $var(x) + ":5060";<BR>á á á á á á á ááif ( !match_dialog() )<BR>
á á á á á á á áá{<BR>á á á á á á á á á á á áácreate_dialog();<BR>á á á á á á á áá}<BR>á á á á á á á á $dlg_val(host) = $du;<BR>á á á áá} else {<BR>
<br><BR>
á á á á á á á á if ( !match_dialog() )<BR>
á á á á á á á á {<BR>
á á á á á á á á á á á á create_dialog();<BR>
á á á á á á á á }<BR>
á á á á á á á á $dlg_val(caller) = $fU;<BR>
á á á á á á á á $dlg_val(callee) = $rU;<BR>á á á á }<br><BR><br><BR><br><BR>You will need to somehow make this work for your setup but hopefully this shows you what you are looking for.<BR>
</div></div><div class="ecxgmail_extra"><br><br><div class="ecxgmail_quote">On Mon, Dec 16, 2013 at 7:31 PM, Jeff Pyle <span dir="ltr"><<a href="mailto:jpyle@fidelityvoice.com" target="_blank">jpyle@fidelityvoice.com</a>></span> wrote:<br>
<blockquote class="ecxgmail_quote" style="border-left:1px #ccc solid;padding-left:1ex;"><div dir="ltr"><div class="ecxgmail_extra"><div><div dir="ltr"><div>Jorge,</div><div><br></div><div>This is a function of Asterisk, not Opensips. áThis page may help you:</div>
<div>áá<a href="http://www.voztovoice.org/?q=node/350" target="_blank">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="ecxgmail_quote"><div><div class="h5">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></div>
</div><blockquote class="ecxgmail_quote" style="border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex;"><div><div class="h5">
<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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<br></blockquote></div><br><br clear="all"><div><br></div>-- <br>--<br>*--*--*--*--*--*<br>Duane<br>*--*--*--*--*--*<br>--
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