Hi Aamir,<div>I remember on one of my recent projects using OpenSIPS and Asterisk/FreeSWITCH even SEMS I required to do exactly what you are asking i.e get DTMF from the caller and then the Media-Server disappears from the call. Like respected Olle suggested, I also tried using B2BUA module, tried using the REFER things from asterisk and FreeSWITCH both but it did nothing what I expected. </div>
<div><br></div><div>DTMF collection in session progress is good thing, you definitely need to tweak the UACs SIP timers nor to get expired from 183 event while collecting DTMFs. I wonder if you can control the UACs.</div>
<div><br></div><div>What I think you should do is keep the media-server in SIP signalling path only once after the IVR has been played and DTMF has been collected. </div><div>Thats what I think about the scenario.</div><div>
<br></div><div>Regards</div><div>Sammy</div><div><br><br><div class="gmail_quote">On Mon, Jul 2, 2012 at 7:36 PM, Olle E. Johansson <span dir="ltr"><<a href="mailto:oej@edvina.net" target="_blank">oej@edvina.net</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><br>
2 jul 2012 kl. 16:08 skrev aamir chougule:<br>
<div class="im"><br>
> Hi Olle,<br>
><br>
> 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.<br>
</div>There's always a way... FedEx does this in the US - running an IVR in early media. Now, support of sending DTMF before answering a call is something poorly specified and you will have a hard time with interoperability.<br>
<br>
Cheers,<br>
/O<br>
<div class="im HOEnZb">><br>
> Thanking you in anticipation.<br>
><br>
> Regards,<br>
><br>
> Aamir Chougule<br>
> Cell: 09167989111<br>
><br>
</div><div class="im HOEnZb">> From: Olle E. Johansson <<a href="mailto:oej@edvina.net">oej@edvina.net</a>><br>
> To: aamir chougule <<a href="mailto:aamir_ryu@yahoo.com">aamir_ryu@yahoo.com</a>>; OpenSIPS users mailling list <<a href="mailto:users@lists.opensips.org">users@lists.opensips.org</a>><br>
> Sent: Monday, 2 July 2012 7:08 PM<br>
> Subject: Re: [OpenSIPS-Users] OpenSIPS & Asterisk Integration in a new way<br>
><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>
<br>
<br>
</div><div class="HOEnZb"><div class="h5">_______________________________________________<br>
Users mailing list<br>
<a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><br>
<a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br>
</div></div></blockquote></div><br></div>