Strange Behavior

I have Asterisk 11 with Freepbx 2.11.65 setup with a digium td410 analog fxo board connected to one analog pots line. I also have Google voice connected and working correctly. I have incoming route setup to forward any pstn call to a ring group and another route to forward any incoming Google call to a specific extension. What is happening is that when a call comes in on the Google voice line my hardware seems to send a ring signal to my analog phone, it will ring once then asterisk detects this as if it were a call from the outside and send a ring signal to the ring group. This only last two rings on the Sip extensions but is very annoying. The route works correctly and the call is sent to the correct extension. In the cdr report it shows the incoming call as two separate calls.
How can I prevent this from occurring ? One further note, if the POTS line is in use this does not occur and everything is routed correctly.

Here is some log info:

[2014-08-20 17:25:38] VERBOSE[26105][C-00000010] pbx.c: – Executing [s@macro-dial-one:35] Set(“Motif/+1305XXXXXXX-029d”, “__CWIGNORE=”) in new stack
[2014-08-20 17:25:38] VERBOSE[26105][C-00000010] pbx.c: – Executing [s@macro-dial-one:36] Set(“Motif/+1305XXXXXXX-029d”, “__KEEPCID=TRUE”) in new stack
[2014-08-20 17:25:38] VERBOSE[26105][C-00000010] pbx.c: – Executing [s@macro-dial-one:37] GotoIf(“Motif/+1305XXXXXXX-029d”, “0?usegoto,1”) in new stack
[2014-08-20 17:25:38] VERBOSE[26105][C-00000010] pbx.c: – Executing [s@macro-dial-one:38] GotoIf(“Motif/+1305XXXXXXX-029d”, “1?godial”) in new stack
[2014-08-20 17:25:38] VERBOSE[26105][C-00000010] pbx.c: – Goto (macro-dial-one,s,43)
[2014-08-20 17:25:38] VERBOSE[26105][C-00000010] pbx.c: – Executing [s@macro-dial-one:43] Dial(“Motif/+1305XXXXXXX-029d”, “SCCP/5000,30,TrWwt”) in new stack
[2014-08-20 17:25:38] VERBOSE[26105][C-00000010] app_dial.c: – Called SCCP/5000
[2014-08-20 17:25:38] VERBOSE[26105][C-00000010] app_dial.c: – SCCP/5000-00000010 is ringing
[2014-08-20 17:25:38] VERBOSE[26199][C-00000011] sig_analog.c: – Starting simple switch on ‘DAHDI/1-1’
[2014-08-20 17:25:38] VERBOSE[26199][C-00000011] pbx.c: – Executing [s@from-analog:1] NoOp(“DAHDI/1-1”, "Entering from-dahdi with DID == ") in new stack
[2014-08-20 17:25:38] VERBOSE[26199][C-00000011] pbx.c: – Executing [s@from-analog:2] Ringing(“DAHDI/1-1”, “”) in new stack
[2014-08-20 17:25:38] VERBOSE[26199][C-00000011] pbx.c: – Executing [s@from-analog:3] Set(“DAHDI/1-1”, “DID=s”) in new stack
[2014-08-20 17:25:38] VERBOSE[26199][C-00000011] pbx.c: – Executing [s@from-analog:4] NoOp(“DAHDI/1-1”, “DID is now s”) in new stack
[2014-08-20 17:25:38] VERBOSE[26199][C-00000011] pbx.c: – Executing [s@from-analog:5] GotoIf(“DAHDI/1-1”, “1?dahdiok:checkzap”) in new stack
[2014-08-20 17:25:38] VERBOSE[26199][C-00000011] pbx.c: – Goto (from-analog,s,9)
[2014-08-20 17:25:38] VERBOSE[26199][C-00000011] pbx.c: – Executing [s@from-analog:9] NoOp(“DAHDI/1-1”, “Is a DAHDi Channel”) in new stack
[2014-08-20 17:25:38] VERBOSE[26199][C-00000011] pbx.c: – Executing [s@from-analog:10] Set(“DAHDI/1-1”, “CHAN=1-1”) in new stack
[2014-08-20 17:25:38] VERBOSE[26199][C-00000011] pbx.c: – Executing [s@from-analog:11] Set(“DAHDI/1-1”, “CHAN=1”) in new stack
[2014-08-20 17:25:38] VERBOSE[26199][C-00000011] pbx.c: – Executing [s@from-analog:12] Macro(“DAHDI/1-1”, “from-dahdi-1,s,1”) in new stack

I’d suspect a wiring fault or a problem with the Digium cards. In the latter case, especially, you should be using Digium commercial support, not the peer support for the open source software.

Thank you for responding. The hardware is functioning properly. If you notice in the log asterisk sends out the ring signal.

It sends it out but it is getting looped back somewhere in the hardware, e.g. as the result of cross talk.

The software should not be sending a ring signal out on dahdi 1/1 on an incoming googlevoice (ip) call. cross talk occurs only adjacent channels in a wire trunk or card, what does that have to do with the issue I am describing ? dahdi 1/1 is connected to the pots line.

Why shouldn’t it be sending ringing current? There is nothing in the information you have provided that would lead me to expect that it wouldn’t do so.

Cross talk is most likely due to incorrect wiring, with a fault in the line card as a much less likely possibility. Line card faults should definitely be addressed through the supplier’s commercial support channels. Digium do not provide support on the open source support forums.

dahidi 1/1 is an FXO port. It is not in the ring group, and is not the extension that the call is sent to.

FXO ports need FXS signalling.

That is how its configured…

digium.com/en/support/telephony-cards