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.
Thanking you in anticipation.
Regards,
Aamir Chougule
Cell: 09167989111
________________________________
From: Olle E. Johansson <[email protected]>
To: aamir chougule <[email protected]>; OpenSIPS users mailling list
<[email protected]>
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
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