<html><body><div style="color:#000; background-color:#fff; font-family:times new roman, new york, times, serif;font-size:12pt"><div><span>Hi Olle,</span></div><div><br><span></span></div><div><span>Thanks for the genuine suggestion and I really appreciate your answer. I understand the complications now after hearing the answers but is there a way before answering a call fetching the digits and then sending the digits back to the opensips and proxy it through the opensips to the carrier. I know for IVR answering a call is a must, BUT is there an option to collect digits that will be dialed by the customer and send to the opensips for the call initiated and then billing will be a easier thing to do.</span></div><div><br><span></span></div><div><span>Thanking you in anticipation.</span></div><div> </div><div style="font-family:times new roman, new york, times, serif;font-weight:bold;color:rgb(3, 61, 33);">Regards,<br><br>Aamir Chougule<span
class="tab"></span></div><div style="color:rgb(0, 0, 0);font-size:13px;font-family:arial, helvetica, clean, sans-serif;background-color:transparent;font-style:normal;"><span class="tab"><span style="font-family:times new roman, new york, times, serif;font-weight:bold;text-decoration:underline;color:rgb(3, 61, 33);">Cell: 09167989111</span><br></span></div><div><br></div> <div style="font-family: times new roman, new york, times, serif; font-size: 12pt;"> <div style="font-family: times new roman, new york, times, serif; font-size: 12pt;"> <div dir="ltr"> <font face="Arial" size="2"> <hr size="1"> <b><span style="font-weight:bold;">From:</span></b> Olle E. Johansson <oej@edvina.net><br> <b><span style="font-weight: bold;">To:</span></b> aamir chougule <aamir_ryu@yahoo.com>; OpenSIPS users mailling list <users@lists.opensips.org> <br> <b><span style="font-weight: bold;">Sent:</span></b> Monday, 2 July 2012 7:08 PM<br> <b><span
style="font-weight: bold;">Subject:</span></b> Re: [OpenSIPS-Users] OpenSIPS & Asterisk Integration in a new way<br> </font> </div> <br><br>2 jul 2012 kl. 13:34 skrev aamir chougule:<br><br>> Wanted Scenario:<br>> <br>> Calls comes in to OpenSIPS server ==> Authentication & Proxying part will be done by OpenSIPS ==> Call is relayed to Asterisk Server ==> Asterisk Server provides the IVR services to fetch the number from the customer ==> Asterisk passes on the fetched number to the OpenSIPS Server ==> OpenSIPS server relays the call to the carrier according to the LCR<br>> <br>THis will be hard to do, OpenSIPS is in general a proxy and you can't transfer a call to a proxy. <br>Before answering you could use the transfer() application in the Asterisk dialplan to send a SIP 302 redirect and the proxy could forward the call.<br><br>In this case, you are actually answering the call in order to perform the IVR. This
means that you have to send a <br>SIP REFER message, which the proxy can't handle. It goes all the way to the caller who then issues another INVITE.<br><br>I don't know what you can do with the OpenSIPS b2bua module, maybe that module can handle a REFER and help you.<br>In Asterisk, you can issue a REFER to transfer the call with the transfer() dialplan application too. <br><br>/O<br><br> </div> </div> </div></body></html>