I configured the voice trunk (SIP) as follows
voice trunk T01 type sip
grammar request-uri host domain
grammar from host domain
grammar to host domain
match dnis "NXXNXXXXXX" substitute "1NXXNXXXXXX"
sip-server primary #registrar# udp 5065
registrar primary #registrar# udp 5065
authentication username "#SIP username#" password "#SIP Password#"
When placing an inbound call, the adTran is looking at the SIP URI instead of the TO header, however the adTran returns a 404 not found because it is looking at the SIP URI instead of the TO header coming from the SIP server.
Is there any setting that I need to change to have the adTran look at the TO header coming from the SIP server to route the call?
I dont want to register every single phone number that comes to the adTran.
There are some pretty powerful SIP header manipulation tweaks available. See Manipulating SIP Headers and Messages in AOS for exact syntax.
Another option would be to configure a registered trunk with a pilot number and have the DIDs come over that trunk (which I think may be exactly what you're attempting to do).