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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
Current Document Icon 8. Line 1 sub tab
Document Icon 9. PSTN Line sub tab
Document Icon 10. Conclusion

The Line 1 sub tab

Before updating any more configuration parameters we'll need an actual SIP provider account. If you already have a SIP account obviously you can skip this bit. There are many organisations offering SIP VoIP services. I don't really want to endorse any of the major providers, so I'll leave that for you to work out. I chose a company called DrayTel on the basis of a recommendation. Their prices seem fair by comparison with other providers, and they offer all the services that I need.

With DrayTel, you pay nothing for the basic VoIP account which allows you to communicate with other SIP users across the internet for free. It's pretty much like free e-mail signup. Then, when you buy the first £10 of call time you are allocated a telephone number and you can choose whatever UK area code you want. You can use VoIP to dial out on that number and people can contact you on it.

DrayTel offer a customer referral scheme. When you buy your first £10 of call time, both the referrer and referee get an additional £2 of call time. You just have to enter the referring SIP user number. Mine is "8243002", enter my number if you buy their services, and get your free call time (nudge - nudge) :).

Once you've registered and when you get your details, you'll have a number in the form, and a password. You may or may not have a telephone number associated with the account, but it doesn't actually affect setting up the SPA3102. The association of the number occurs with at and with the SIP service provider. The next task is to pass these details to the SPA3102. This is done in the setup below;

Line Enable: Yes

Proxy and Registration
Proxy: This is the name of your SIP provider i.e.
Outbound Proxy: If you are using NAT behind a router, you can put the name of your SIP provider's Outbound Proxy here
Use Outbound Proxy: Yes If you provide a name in the outbound proxy box
Register: Yes

Subscriber Information
Display Name: This is the full name of your SIP/VoIP account i.e.
User ID: This is the numeric part of your user id i.e. 1234567
Password: Your SIP/VoIP pasword goes here
Use Auth ID: Yes
Auth ID: This is the numeric part of your user id i.e. 1234567 (again)

As far as I understand this, the details above represent your "central", dial out, VoIP account. When you want to receive calls from a variety of different telephone numbers, each would point at that central account. Typically these multiple "dial in" numbers are available in most countries, and are either very cheap or free. This is distinct from the section below where you can specify a variety of different "voice gateways". These are "dial out" SIP accounts that dial out at a physical phone number. Obviously these must be paid for somehow, and thus their registration is far more formal. If, like me, you only have one SIP account then your Gateway 1 (gw1) is the same as the central SIP account. Details for Gateway 1 follow;

Gateway Accounts
Gateway 1: This is the full name of your SIP/VoIP account i.e. (again)
GW1 NAT Mapping Enable: Yes
GW1 Auth ID: This is the numeric part of your user id i.e. 1234567 (again)
GW1 Password: Your SIP/VoIP pasword goes here

Since I have only the one account it does not matter to me, but I have a feeling that there may be a problem with the way this works. My SIP provider offers a single telephone number which supports both incoming and outgoing calls. This phone number is associated with a single SIP account. They are not obviously separable. I am not sure how one would redirect the incoming number they give me to a different SIP address. If I chose a different primary SIP provider, I imagine this to be a specific problem. I suspect that this is not uncommon provider offering. Nevertheless I'm not sure if that is a problem with the SPA3102, or my SIP provider. Either way it may be difficult to get multiple telephone numbers with the intention of both making and receiving calls on each number, all from a single telephone handset. From the perspective of the SPA3102, presumably the trick is to find the SIP providers that can do this.

Whilst on the subject of Voice Gateways, it is also worth mentioning that because the SPA3102 has an FXO port, Gateway 0 (gw0) is used to reference outgoing calls to the PSTN. From the perspective of the SPA3102 this distinction between gw0 and gw1 is important because it allows one to specify targets in dial plans. Dial plans are like filters used during dialling. Any number of digit sequences can be specified. When you dial a number on the handset, the SPA3102 tries to match each sequence with the number dialled. Each sequence is associated with a Voice Gateway, and when a match is made the outgoing number is invoked (dialled) against the selected Voice Gateway.

A basic dial plan is given below. It allows you to dial 5 to get an analogue line or by default will route to VoIP. It is straightforward to reverse this procedure, i.e. dial 5 to get VoIP. The digit 5 was chosen rather than the more natural 9, since the digit 9 conflicts with the voicemail functions of my SIP provider. These voicemail functions all begin with 9 and the use of the digit 9 for selecting an analogue line renders the voicemail functions impossible to use. If you want to experiment with dial plans, a syntax reference is included in the folder at the top left of this page.

VoIP Fallback To PSTN
Auto PSTN Fallback: Yes Specifying yes means the PSTN line will be used if the internet connection fails. It could be expensive if you dial to the outback and don't realise the internet is not involved

Dial Plan
Dial Plan: (999<:@gw0>|<5:>xx.<:@gw0>|xx.<:@gw1>|x) This will always direct 999 calls to the analogue line. Dial 5 for an analogue line, by default use VoIP. Allow a single digit for speed dial.
Emergency Number: 999
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Last modified: SolFlu  Fri, 16 Oct 2009 15:41:01 GMT