Asterisk and SIP-T/SIP-I

Does Asterisk support SIP-T/SIP-I call flow natively? My limited understanding to this is the answer is no.

It is my understanding SIP-T flow is not quite the same as SIP, some ISUP is sprinkled in and according to the RFC (rfc3372) SIP-T messages look a little different and would require some MGC to handle message translation.

Could Asterisk be configured to handle SIP-T/SIP-I?

That is if Asterisk were built to handle those messages could that be done solely in the dialplan, or would that require a channel driver, something like or or some other program written to handle the flow, translate it, and pass it on…then that program is called in the dialplan.

Thanks for any input. I am fairly new to SIP and have never encountered SIP-T/SIP-I before so my understanding of it is zero except for what I have read over the last couple of days. I would be grateful to discuss this with anyone. Thanks.

giving a bump for any input.

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