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A guide to controlling tone, cadence and dial plans on the SPA3102.

Document Icon 1. Overview
Document Icon 2. What's in the box?
Document Icon 3. Scope of instruction
Document Icon 4. Gaining control
Document Icon 5. Voice tab
Document Icon 6. SIP sub tab
Document Icon 7. Regional sub tab
Document Icon 8. Line 1 sub tab
Document Icon 9. PSTN Line sub tab
Current Document Icon 10. Conclusion


Having been tenacious, and got this far, you should now get the reward for all the effort. You should now be able to disconnect the phone and insert the SPA3102 between the exchange and the handset. Do make sure that you connect the exchange wall socket to the line port and the phone handset to the phone port. Getting this the wrong way around might not be too good. In a battle between the exchange and the SPA3102, I'd put money on the exchange not noticing the fight.

Once you are so connected, you should be able to hear a dial tone when you lift the handset. It is worth remembering that you are hearing the SPA3102 dial tone, and not that of the exchange. This is probably a good time to get your mobile handy and use it to test the operation of the new setup. This can be easier to do, if you have someone to help you, but you can do it alone just as well.

If your phone handset has the capability to dial a number first, and then lift the handset, it's a good way to check the operation of the analogue phone line. Dial 5 and follow up with your mobile number. Put the handset to your ear before pushing the off hook button. You should hear the dial tone, and then the DTMF tones beeping away. All of that is the dialogue between your phone and the SPA3102. Then there will be a click as it decides to dial out on the analogue line. You should then hear the same sequence of DTMF tones, but quieter. This is the SPA3102 dialing your mobile number out to the exchange. If the mobile actually rings, your're winning!

Overall, there are five tests you'll want to make against the four call routes. The table below illustrates these tests, you might even want to print the table out;

 Handset » MobileMobile » Handset
Phone Rings?    
Audio?Handset » Mobile    
Mobile » Handset    
Ring Out?N/AN/A  

Each box should get a good result. Until you check them you can't know for sure. The Phone Rings test just ensures that SIP is capable of making the route across the internet, and that the SPA3102 can dial and be dialed on the PSTN. The audio test makes sure that audio is actually making it through the firewall, and of inteligable quality in both directions. The echo test allows you to determine if there is echo present in a particular situation. Finally the Ring Out test is only valid when calling in to the SPA3102 from the mobile. It aims to ensure that there is no security compromise in the circumstance where no-one answers the phone. In this last test you are aiming to get a disconnect, and not any kind of tone, when the call fails to be answered.

If you have problems this list will help you to remember the specific scenario where the problem is. That will certainly help since there are potentially 22 different kinds of failure. There is nothing worse than trying to fix a fault when you can't remember what it was!

In truth, echo is likely to be a problem on the handset for PSTN calls. The best way to resolve these is by fiddling with the echo canceller, and the FXS Port gains. Some of the parameters involved are;

Voice » Regional » Miscellaneous » FXS Port Input Gain:
Voice » PSTN Line » International Control » SPA To PSTN Gain
Voice » PSTN Line » Audio Configuration » Echo Canc Enable:
Voice » PSTN Line » Audio Configuration » Echo Canc Adapt Enable:
Voice » PSTN Line » Audio Configuration » Echo Supp Enable:

Be aware that if you push some parameters too hard you will begin to distort the output. This will sound even more unpleasant than any echo and should be avoided. In the case of the two gain parameters, you probably want to err on the side of smaller numbers than larger ones, to avoid echo.

As a final word, I'll mention the capability to forward callers from the VoIP port on to the PSTN. The reverse procedure is very similar in nature. The User1 and PSTN User pages are for these purposes respectively. To allow a VoIP caller to call out on the PSTN, on the User1 sub-tab simply set "Cfwd No Ans Delay:" to the number of seconds you wish the phone to ring before forwarding. Set "Cfwd No Ans Dest:" to "gw0" (without the quotes). It is advisable to set a password, and it will only work if you specify a dial plan. Be mindful that this capability may interfere with your voicemail functionality.

Happy calling!

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Last modified: SolFlu  Mon, 21 Sep 2009 02:05:32 GMT