[OpenSIPS-Users] OpenSIPS & Asterisk Integration in a new way

[Olle E. Johansson]

2 jul 2012 kl. 16:08 skrev aamir chougule:

> Hi Olle,
> 
> 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.
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.

Cheers,
/O
> 
> Thanking you in anticipation.
>  
> Regards,
> 
> Aamir Chougule
> Cell: 09167989111
> 
> From: Olle E. Johansson <oej at edvina.net>
> To: aamir chougule <aamir_ryu at yahoo.com>; OpenSIPS users mailling list <users at lists.opensips.org> 
> Sent: Monday, 2 July 2012 7:08 PM
> Subject: Re: [OpenSIPS-Users] OpenSIPS & Asterisk Integration in a new way
> 
> 
> 2 jul 2012 kl. 13:34 skrev aamir chougule:
> 
> > Wanted Scenario:
> > 
> > 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
> > 
> THis will be hard to do, OpenSIPS is in general a proxy and you can't transfer a call to a proxy. 
> 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.
> 
> In this case, you are actually answering the call in order to perform the IVR. This means that you have to send a 
> SIP REFER message, which the proxy can't handle. It goes all the way to the caller who then issues another INVITE.
> 
> I don't know what you can do with the OpenSIPS b2bua module, maybe that module can handle a REFER and help you.
> In Asterisk, you can issue a REFER to transfer the call with the transfer() dialplan application too. 
> 
> /O
>