Hello
After a bit of fiddling, I got to the web interface and can more or less successfully receive a call through an Asterisk server using this SIP gateway.
I have the following questions, though:
- HT488 doesn't detect that the remote end has hung up: What should I use for France as regional settings in the FXO Port section? Here are the default settings that show up in the web interface: PSTN AC Termination: 320 Ohm + (1050 Ohm || 230 nF)) PSTN Disconnect Tone: Frequency: f1 480 f2 620 PSTN Disconnect Tone Cadence: All 0's PSTN Silence Timeout : 60
- No CID is returned, even though it works OK if I plug a caller ID-capable modem on the phone line. HT488 doesn't handle CID, or is it linked to regional settings above?
- How to secure access to PSTN line so that unauthorized users from the Net don't use my PSTN line?
Thank you.