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

[SamyGo]

Hi Aamir,
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.

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.

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.
Thats what I think about the scenario.

Regards
Sammy


On Mon, Jul 2, 2012 at 7:36 PM, Olle E. Johansson <oej at edvina.net> wrote:

>
> 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
> >
>
>
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