Line (or FXO aka PSTN) is meant to connect to a Telco, or PSTN. Does that mean that it answers if it hears a ring?
And then does what?
The SPA3102 FXO port will hear a ring and initially ring the phone attached to the SPA3102 waiting for an answer and then, if answered, connecting the phone to the pstn line. This assumes the "Ring Thru Line 1" setting is set to Yes.
After the expiration of the "PSTN Answer Delay", assuming the call was not answered by the phone, the SPA3102 logic shifts to the "PSTN-to-VoIP Gateway" logic. Depending on the configuration the SPA3102 will return a dial tone or automatically dial a voip call. The outbound sip call uses the sip configuration setup on the PSTN Line Tab.
Client has switched their phone provider to a cloud based voip service provider.
They have a speakerphone system at their door that had been connected to their previous phone system.
Upon investigation (good old trial and error), I have learned that the speakerphone sends a ring signal, when the button is pressed.
So, I'm guessing that should be plugged into the FXO/PSTN port.
Assuming you have a two-wire cable from the Speakerphone unit to the SPA3102, the SPA3102 FXO port will detect ringing on the line and will ring the phone attached to the SPA3102. The Speakerphone unit will need to provide the both the ringing voltage and the talk voltage to carry the analog voice signal.
You need to have the "Ring Thru Line 1" feature enabled on the PSTN Line Tab and have a PSTN Answer Delay setup for a significant number of seconds to allow ringing of the phone. Both of these are default settings.
If you do not wish the phone attached to the SPA3102 to ring, but wish the SPA3102 to make an outboud sip call, then you would disable the "Ring Thru Line 1" feature and reduce the PSTN Answer Delay to 0. Then you would need to setup an appropriate configuration on the PSTN Line Tab and setup a dial plan to dial the outgoing sip call.
It can also be setup to initially ring the phone attached to the SPA3102 and then if not answered switch to the pstn-to-voip gateway and dial an outgoing sip call.
Now I've already configured the Phone Port (FXS) so it's a hot line, take the connected phone off hook and it dials,
but I found to do that with a p0 code... and on here i've seen s0 codes...
What's the difference?
I'm not sure what you are doing with the FXS port.
The "P" parameter stands for pause. It is often called a Dial Plan Timer. You can see some examples of its use in the Cisco ATA Administration Guide....http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/ata/administration/guide/ATA_AG_v3_NC-WEB.pdf
The "S" parameter is for the Short Interdigit Timer. When used and set to zero it causes immediate dialing of whatever has been dialed without waiting for any additional digits. In some circumstances the P0 and S0 can be used for the same thing.
