Grandstream HT813 PSTN passthrough and sip invice

FreePBX Endpoints
1

Hi all.
I’m evaluating the HT813. I need just a “simple” thing: being able to ALWAYS route mechanically/physically/electrically a PSTN call from the FSX to the FXO port (the lifeline support) and, at the same time, also call a sip extension (just to get the caller-id).

Most of the time, the SIP extension won’t answer, the call would be answered from the analog phone, the sip only get the caller-id to trigger some action but sometimes, based on the caller-id, it could answer, and when this happens, the analog phone should stop ringing. In other words, the first that answer, should stop ringing the other.

Is this possible ? If a power outage should happens, the analog line should still work even with the grandstream powered off.

2

Don’t connect the FXS port at all. Connect the FXO port in parallel with the direct connection to the phone.

3

doing so i wont be able to direct call to the analog phone originating from the VoIP server

4

In that case you will need to wire a relay, probably best with break before make contacts, and connect it to the same power as the ATA. I’m not sure if you will need to fully isolate both sides from the ATA.