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 chan_sip_t.so or chan_sip_i.so? 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.