[Asterisk-Dev] T.38 pass-through / T.38 support

[Kevin P. Fleming]

Steve Underwood wrote:

> through *. What I assume happens is the ATA calls into *. * gives the 
> above message. A G.711 call goes through to the Cisco. A re-invite 
> connects those 2 directly, bypassing *. They chat amongst themselves, 
> renegotiate, and talk UDPTL directly.

Jumping in late...

Even if this scenario is true, they cannot 'chat amongst themselves'. A 
SIP reinvite moves _only_ the media path, not the signaling. Any 
renegotiation of the media stream would still go through Asterisk, if 
that's how the original call was placed.

I too am wondering just how Matthew Boehm's system is working; as of 
this point, a T.38 call of any form (RTP or UDPTL) can't be handled by 
chan_sip, and an existing call can't turn into T.38 either.