SPA3102 PSTN to VOIP

[peteg]

Hi folks,
We're moving our office phone system over to VOIP (using voipfone.co.uk as our provider). I've purchased an SPA3102 to divert incoming pstn calls on our old number to our hosted voip pbx.
It's on our network, behind our router. We use 2 class C networks behind our router (192.168.0.x for "public facing" and 192.168.1.x for inward facing). I connected the LAN port to our 192.168.1.x network and configured it as instructed at http://linksys.custhelp.com/cgi-bin/...=8787043103B12. At this point it would make a sip registration, so I also connected the WAN port to our 192.168.0.x network, and it now registers OK. BUT, the call forwarding doesn't work - the pstn call is not picked up. Interestingly, when I look at the "Info" page on the spa3102 in this condition, under "PSTN Line Status" it shows:

VoIP Peer Number:XXXXXXXX@127.0.0.1:5060

...ie with a local host ip - is this correct? If so, can anyone point me in the right direction to get this working?

[hwittenb]

The 127.0.0.1 is a call inside the SPA3102 from the PSTN port to the phone attached to the FXS port on the SPA, so called "Ring Thru to Line 1". This goes on for the duration of the PSTN Answer Delay to allow a call to be answered by the handset. If you do not want this, set Ring Thru to Line 1 to No, and set the PSTN Answer Delay to 0 or 1 or 2 depending if you want to capture an incoming Caller ID. Then the pstn-to-voip gateway will answer the phone and you should have a dial plan setup for the pstn-to-voip gatway that will automatically send the call to the appropriate voip provider number or sip uri.

[peteg]

Hi - sorry for the delay in my reponse. Yes, what you makes sense - I'll take another look.

Thanks!