SPA3102 - No Line Voltage

I'm having a problem with getting my SPA3102 to ring on my PSTN. I plugged it from the line jack into my wall jack. Then i plugged a phone in another wall jack. There's no dial tone for the phone and the info page is showing I don't have any line voltage.
Does anyone have any ideas how to fix this? Here's what the info page says below...
Hook State: On
Line Voltage: 0 (V)
Loop Current: 0.0 (mA)
Registration State: Registered
Also, i can call my phone ok if i hook it directly to the 3102 phone jack and setup Line 1.
Thanks in advance.

The "Line" jack on the SPA3102 is an "FXO" interface. The "Phone" jack on the SPA3102 is an "FXS" interface. They are not the same. The "FXO" interface is to attach the SPA3102 to a working pstn line that is also attached to your phone company. The "FXS" interface is to attach the SPA3102 to station equipment such as a phone handset.
If you have no pstn phone service you should not be using the "FXO" or line interface. If you wish to ring additional telephones in the house from an incoming voip call to the SPA3102 they should be attached to the "FXS" or phone interface, however the phones should not also be attached to a working pstn phone line.

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I'm trying to set this device up in this specific scenario:
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Spa3102 would not forward a voip call to pstn line

Good morning.
I've done the implementation provided here http://community.linksys.com/t5/VoIP-Adapters/SPA-3102-and-softphone-to-
make-calls-via-pstn-line/td-p/326390 .
It is a way to use for outgoing calls a given pstn line from anywhere I have internet (voip to pstn).
The spa3102 is connected to a router (with an active DHCP server and ip 192.168.1.1) from where it takes the internal
ip (192.168.1.3).On the same network is also a computer , connected to the router ( with ip 192.168.1.2). The spa3102
is set to bridge mode and thus inactivates the function of the router (on SPA3102), and it functions as a simple
network device . I have done port forwarding (from the router) to 192.168.1.3 (SPA3102) for the port 5061 (PSTN
LINE) ( but for 5060 for the LINE 1 also). I want to make calls from a voip softphone (x-lite 4) to the SPA 3102 and
this to forward the voip calls to PSTN line to which it is connected. In x-lite the SPA3102 is set as a proxy so that
i can type the phone number I want to call without being followed by the SPA3102's ip each time ( eg on x-lite I
give call number 2101111111 instead of 2101111111 @ wanip: 5061 where wanip is the external ip of the router).
When x-lite is running on the computer that is on the same network with the SPA3102 everything works as expected. A
voip call is made from x-lite ( using as a proxy the wanip everytime, or even for test purposes the dyndns domain
that i set up for this reason), this call is passese PSTN line and the phone of the called party rings . At x-lite
COMES indication "call established ".
The problem occurs when I do the same procedure from x-lite installed on a computer belonging to another network (
e.g. in another building with its own internet connection , own router, own computer , etc. ) . Always using the
wanip the x-lite makes the voip call to the SPA3102, writes "call established" ( meaning it connected to SPA3102) but
never routed the call to the called party ( the SPA3102 did not forward voip calls it receives to the PSTN line ) .
Trying to find what 's wrong I've tried to disable all firewalls (soft and hard from all involved machines ) . The
behavior is the same either the computer that makes the successful calls is connected to the network directly to the
router or through the port "ethernet" on the SPA3102.
What is the difference in these two voip calls to the SPA3102 and the one " triggers " it to forward the call to
PSTN line and the other does not ?
Thanks now for any ideas you give .

The audio sound problem is more than likely also associated with the overall addressing problem initially encountered. As you may know, using the sip protocol the sip signalling exchanges ip addresses to be used for both the sip signalling and the exchange of rtp sound packets. In addition there is an exchange of port numbers to be used for the exchange of rtp sound packets. The sound is exchanged by two separate streams of packets, one stream in each direction. The result is an ip address and port number for the rtp packets flowing from the SPA3102 to the softphone and a different ip address and port number for the rtp packets flowing from the softphone to the SPA3102.
In your previous posting you mentioned that you "set the minimum EXTernal rtp port at the sip tab". Changing the "EXT RTP Port Min:" is an unusual change to make and in my opinion would only be made in special circumstances. Actually, I ran some tests and I'm not sure exactly what that setting does. In my tests it didn't appear to affect the rtp port number used in a predictable manner.
The common changes to make for audio problems typically would be to setup a STUN server. A STUN server is an external server that echos back to the initial sender the external ip address and port number that the STUN server received with the message received by the server. This allows the sender (SPA3102 or softphone) to determine its external ip address and external port numbers for both the sip signalling and rtp packets.
A STUN server is commonly recommended to be setup with the following settings in the SPA3102:
PSTN Line Tab:
NAT Mapping Enable: Yes
Sip Tab:
Handle VIA received: yes
Handle VIA rport: yes
Insert VIA received: yes
Insert VIA rport: yes
Substitute VIA Addr: yes
Send Resp To Src Port: yes
STUN Enable: yes
STUN Server:
The following web page has a list of "Public STUN Servers"
http://www.voip-info.org/wiki/view/STUN
You are using CounterPath's XLite softphone. stun.counterpath.net is a STUN server on the list.
I see XLite also has a setting to use a STUN server on the "Topology" tab.

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I believe this is a normal behavior according to the admin guide:
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Damon:
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Solved!
Go to Solution.

AK2DM:
Update - solution found
While the Butterworth low-pass filter sort of worked, it was NOT the solution. I spent some time analyzing the rope CT signals by sampling at a very high rate. The AC signal coming off the rope CTs is triangular NOT sinusoidal. The RMS function in LabView assumes a sinusoidal waveform and essentially calculates the RMS as the average of voltage peaks (for each cycle) / sq.rt (2). Since the AC voltage single emanating from a regular split-core CT is basically sinusoidal, the LabView RMS of the signal gives an accurate reading of the voltage signal. Unable to find a RMS function in LabView for a triangular AC voltage signal, I ended up developing my own custom triangular RMS function in LabView. My function analyzes the signal by finding the peak of every cycle within the signal, takes the average of these peaks, then divides this average peak by sq.rt (3). see this link for formulas:
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SPA3102 - No Line Voltage

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