You can achieve this by setting relevant sip flags in the dialplan back and forth.
Mitul On Mar 12, 2014 11:18 PM, "DHAVAL INDRODIYA" <[email protected]> wrote: > > 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
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