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 <o...@edvina.net> > To: aamir chougule <aamir_...@yahoo.com>; OpenSIPS users mailling list > <users@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 > _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
