Looks to me both Line 1 and PSTN Tab are share the same ITSP VoIP A/C
In the example of VoIP-To-PSTN Gateway, when caller call ITSP DID number, how can we know it will connect to Line 1 or PSTN side? If caller being connected to Line 1 I would expect the PSTN gateway won't works because it will just ring the analog phone. Am I missed something?
Another question is that what if one stage dialing in the VoIP-To-PSTN Gateway set to "Yes", can I simply pass a PSTN no. in the SIP URI and get it dial out directly? If the answer is possitive, what would be the format?
I already read page 93 of the ATA Admin Guide, it doesn't provide further details on how to make a One-Stage Dialing. It only explains what is a One-Stage Dialing and mentioned HTTP Digest Authentication can be used but doesn't provide any useful example such as the correct sip uri format for making a OSD call and page 213 also doesn't provide any details on how to construct a HTTP Digest Authentication, etc. So I have to ask for further details here or you can point me to the right document.