SPA232D NAT Mapping Enable/Keep Alive

[FredThompson0340]

I am using my SPA232D via my home network router, where it has a reserved IP address 192.168.1.X. My router receives a dynamic IP address from my ISP.

I am using PSTN and one SIP account on my system with associated directing dialling plan.

Should I enable NAT mapping enable & NAT keep alive on SIP & PSTN?

I am having some issues whereby sometimes when I make a call I only get a busy tone.I recall reading somewhere that using Keep Alive may fix the problem, but I don't fully understand.

Thank you, Fred

[Dan Lukes]

No way to analyze mentioned issues with no log records. Turn on syslog&debug logs and capture them. It will disclose the reason. Captured SIP packets may help as well.

Debug and syslog Messages from SPA1x2 and SPA232D ATA

Generally, NAT keep alives doesn't harm most of time, so turn them on.. According NAT mapping - just try, but it's not necessary most of time and may harm.

[FredThompson0340]

Thank you again Dan.

I have included a (rather long) debug file.

At around 09:50 today, I tried to dial '1907' a number of times, but only got a dial tone.

According to my dial plan, it should have been sent via @gw0

Can you help ?

Regards, Fred

[Dan Lukes]

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At around 09:50 today, I tried to dial '1907' a number of times, but only got a dial tone.

According to my dial plan, it should have been sent via @gw0

Lets go. I will quote & comment most important records:

Cisco [0]Off Hook (you picked handset off the cradle)

Cisco NEW_CALL_STATE(), call 0: old state = CC_CST_IDLE, new state CC_CST_DIALING (device is waiting for digits dialed)
Cisco SLIC_startTone 1 (you got dial tone)

Cisco line:0 DTMFON event received digit 1 (first digit received)

Cisco SLIC_stopTone (dial tone turned off)

Cisco line:0 DTMFON event received digit 9 (second digit received)

Cisco line:0 DTMFON event received digit 0 (third digit received)

Cisco line:0 DTMFON event received digit 7 (fourth digit received)

Cisco NEW_CALL_STATE(), call 0: old state = CC_CST_DIALING, new state CC_CST_CALLING (target number collected, device is going to call)

Cisco ++++ called number = 1907

Cisco FXO:Off Hook (FXO line is going off-hook)

Cisco _uchPlayFXODTMF(), Play digit 1

Cisco _uchPlayFXODTMF(), Play digit 9

Cisco _uchPlayFXODTMF(), Play digit 0

Cisco _uchPlayFXODTMF(), Play digit 7 (digits 1 9 0 7 are DTMF dialed on FXO)

Cisco FXO:dialing out finished (dialing on FXO completed)

Cisco uchConnectEpToNode(), connecting EP FXO to node 1 (call legs are connected)

Cisco [0]On Hook  (you put handset to cradle, call is teared)

I see nothing wrong, you dialed 1907, those digits has been successfully passed to FXO line (so @gw0 routing request has been honored), call has been ended by you. I see nothing wrong with SPA232D setup and operations.

OK. The FXO line you are using here is connected and audio is passed bidirectionally. If you are hearing dial tone, it must be because it's generated by other end of FXO line and passed thru FXO line to you. I don't know what device is on other end of FXO. Dial tone on FXO may be present because (list of reasons is not eshaustive):

• DTMF dialing is transmitted into line so quickly after seizure - line has not been ready yet, DTMF dialing has been missed because of timing issue
• DTMF dialing is not recognized by other end of FXO line - it may be caused by so low or so high volume of DTMF, DTMF signal may be so short, line may be unbalanced (improper impedance setting), there may be so high level of noise on wire, ...
• DTMF dialing of 1907 has been received correctly, dial tone is intended behavior of 1907 line