[asterisk-users] Regarding SIP-T/SIP-I support in Asterisk.

[DHAVAL INDRODIYA]

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 <amit at avhan.com> 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
>
>
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