Thanks Amit, I want following scenario.
INCOMINGCALL ---> MSC (SIP-T) ----> PBX (Asterisk) OUTGOINGCALL ---> PBX (Asterisk) (SIP) to (SIP-T) ---> Aircel MSC I understood that via Dial-plan we can achieve and get extra parameters values. But what about RTP fields as per my analysis ISUP packets are not sending RTP/AVP they are sending multipart data. please correct me if can achieve this functionality. Thanks Dhaval On Wed, Mar 12, 2014 at 6:15 PM, Amit <[email protected]> wrote: > Hi Dhaval, > > Theoretically, Asterisk can support SIP-I / SIP-T. Since protocols provide > additional information and controls, you will not get those benefits. You > will have to write dial plan functions to extract addition information > exposed by SIP-I / SIP-T. > Though, I have not tested it with Asterisk, I have successfully deployed > application on other SIP platforms and interoperability with SIP-I/SIP-T > was not an issue. > > *Regards,* > Amit Patkar > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
