Linksys SPA3102 or SPA3000 Configuration Wizard

This FREE service allows you to easily configure a Sipura/Linksys SPA series adaptor for a number of different services with known good configurations. It will work under the following circumstances:

  • You have a Sipura/Linksys SPA3000 or SPA3102, for other Linksys devices, choose another Device Configuration Wizard;
  • You must have admin-level access to the web interface of the SPA from your current location;
  • The port you wish to configure is unlocked, i.e. not tied to a service provider (if you are asked for a password to configure a port, and your provider has not furnished you with the password, the port will be "locked" and you will be unable to configure it with this wizard);
  • A completely unlocked device must be used for any of the settings that require STUN or if you attempt to change the timezone (i.e. Line 1 cannot be locked by a provider);
  • Your browser supports HTTPS;
  • Your browser allows requested pop-up windows from Voxilla (temporarily disable your pop-up blocker);
  • While not required, we recommend setting the device to factory defaults prior to using the Wizard;
  • You will need an account with SIP userid and password on one of the providers listed in Step 2.

This form will:

  • Collect some basic information to generate your configuration;
  • Open a pop-up window containing a button that will push the requested configuration directly to your device.
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  Step 1 - Enter the IP address of your adaptor  
 
We need to know the IP address of your Sipura adaptor, which you will enter below. To determine your adaptor's IP address, pick up a telephone handset connected to the SPA, dial "****" (that's the star symbol four times), at the voice prompt dial "110#". Your adaptor will read back its IP address to you. Enter the address, including the dots, below, overwriting the "0.0.0.0" entry:
 
     
  Step 2 - Select your Service Provider for Line 1  
 
We need to know which service provider you would like to configure Line 1 of your SPA3000/3102 to utilize. This service will be used for both inbound and outbound calling. Select from one of the following below:
Adam Telefonia IP (Spain)
AdvanceCall (Germany)
Annatel (France)
ATP (Australia - NSW)
ATP (Australia - QLD)
ATP (Australia - SA)
ATP (Australia - VIC)
ATP (Australia - WA)
AXVoice
Bankoi
BusinessCom Networks
Broadeon Communications
BroadVoice Central US
BroadVoice East Coast US
BroadVoice West Coast US
DialDigit
DialPad
Free World Dialup (No NAT)
Free World Dialup (fwdnat proxy)
Free World Dialup (NAT with STUN)
Go2Call
Go2Call (NAT)
Gossiptel
Gossiptel (STUN)
Gradwell
Gradwell (NAT)
iConnectHereMake and Receive Calls
iConnectHere Outbound Only
InPhonex
INX Logical Exchange
iTele (Faroe Islands)
Litespan
Musimi (Denmark)
Mutualphone
Net2Phone.com
Net2Phone.com (NAT)
Nikkom
Nikotel
NuFone
PeopleCall
PhoneGnome
Race Technlogies
Rubicon Networks
Samsung Networks (Korea)
SkyNET Telesystems
Sipgate.de
Sipgate.de (NAT)
Sipgate.co.uk (STUN)
SIPphone (STUN)
Stanaphone
swissipcom (Switzerland)
Telefin (Denmark)
TeleVoIP
Terracall
TrueVoIP
vBuzzer
ViaTalk (Chicago)
ViaTalk (Houston 1)
ViaTalk (Houston 2)
ViaTalk (New Jersey)
ViaTalk (New York)
ViaTalk (Richmond)
ViaTalk (San Jose)
Virgina Broadband
Vozsip
VoiceEclipse
VoicePulse Connect
VoicePulse Open Access
VoIPTalk (a.k.a. TelAppliant)
VoIP User (voipuser.org)
Vonage (softphone account)
Voxee
WorldDialPoint (Australia)
If you are a service provider, support open device configurations, and would like to be included in this configuration wizard, or we are providing incorrect service configuration information, please contact us and we will do our best to correct the situation.
 
     
  Step 3 - Enter your caller ID display name for Line 1  
 
In the box below, enter the name you would like the recipients of your call to see if they have Caller ID.

Note: Even though you may enter a Caller ID display name on this form, not all service providers support the feature. This name will also not show up on outbound PSTN calls.

 
     
  Step 4 - Enter your user ID for Line 1 VoIP Provider  
 
You will need to enter the User ID, as given to you by your service provider, in the box below. Some providers may have you enter something else in this field.

iConnectHere Users: If you registered for a Receiving Calls service, please enter your personal phone number (i.e. 12125551234) here, and proceed to step 5. Otherwise, enter your iConnectHere UserID and proceed to step 6.

 
     
  Step 5 - Enter your Auth ID for Line 1 (not usually required)  
 
If you used the Auth ID field for something other than the SIP username/User ID, e.g. iConnectHere's Receiving Calls service, enter your User ID as assigned by your service provider. Otherwise, leave this blank.

iConnectHere Users: If you registered for a Receiving Calls service, please enter your UserID as assigned by iConnectHere. Otherwise, leave this field blank.

 
     
  Step 6 - Enter your password for Line 1 VoIP Provider  
 
You'll need to enter your password as provided by your service provider.
 
     
  Step 7 - Select your Service Provider for VoIP/PSTN Gateway Services  
 
Please select which VoIP service you would like to use for VoIP to PSTN and PSTN to VoIP Gateway services.

Inbound calls to this VoIP account will be answered by the SPA3000/SPA3102 for the purpose of providing SIP to PSTN Gateway functionality and will not ring the handset attached to Line 1.

Outbound calls using this VoIP account can be made from the handset attached to Line 1 by dialing #1 and then the number, assuming you use the default dial plan specified in Step 12. This account will also be used to make outbound calls when the PSTN-to-VoIP Gateway is utilized.

It is recommended that the account used here be different than the account used in Step 3. Full use of the PSTN to VoIP gateway and VoIP to PSTN gateway requires two different accounts, either with the same provider, or different ones.

If you select the same provider in this step as in Step 2 and the same UserID is used in both Step 3 and Step 8, the Wizard will automatically set "Register: No" and "Make Call Without Reg: Yes" as most Internet telephony service providers do not support multiple registrations. In this configuration, you cannot call in via VoIP and dial out your PSTN line. A second account must be used.

If 'None' is selected below, this means you wish to use your PSTN line only to ring thru calls on the Line 1 handset and make calls out the PSTN line. No VoIP gateway services will be configured. However, for this feature to work, you will need to configure some "bogus" credentials on this line. Proceed to Step 9, choose a bogus UserID, then skip to Step 12, then to step 15, step 17, then finally step 18

Note: The provider used on this line must pass touchtones either via SIP INFO or RFC2833 (AVT). Providers that pass DTMF inband such as BroadVoice and Stanaphone cannot be used here if you are utilizing the VoIP to PSTN gaetway.

Note: If both your Line 1 and PSTN Line VoIP providers require STUN settings, only the Line 1 settings will be used. This should have no impact on your ability to utilize either service since a STUN server is not specific to any particular service provider.

WARNING: Utilizing your VoIP account in a PSTN to VoIP gateway may be against the terms of service of your service provider.

No VoIP Service on PSTN Line
Adam Telefonia IP (Spain)
AdvanceCall (Germany)
Annatel (France)
ATP (Australia - NSW)
ATP (Australia - QLD)
ATP (Australia - SA)
ATP (Australia - VIC)
ATP (Australia - WA)
AXVoice
Bankoi
Broadeon Communications
BusinessCom Networks
BroadVoice Central US
BroadVoice East Coast US
BroadVoice West Coast US
DialDigit
DialPad
Free World Dialup (No NAT)
Free World Dialup (fwdnat proxy)
Free World Dialup (NAT with STUN)
Go2Call
Go2Call (NAT)
Gossiptel
Gossiptel (STUN)
Gradwell
Gradwell (NAT)
iConnectHere Make and Receive Calls
iConnectHere Outbound Only
InPhonex
INX Logical Exchange
iTele (Faroe Islands)
Litespan
Musimi (Denmark)
Mutualphone
Net2Phone.com
Net2Phone.com (NAT)
Nikkom
Nikotel
NuFone
PeopleCall
PhoneGnome
Race Technlogies
Rubicon Networks
Samsung Networks (Korea)
SkyNET Telesystems
Sipgate.de
Sipgate.de (NAT)
Sipgate.co.uk (STUN)
SIPphone (STUN)
Stanaphone
swissipcom (Switzerland)
Telefin (Denmark)
TeleVoIP
Terracall
TrueVoIP
vBuzzer
ViaTalk (Chicago)
ViaTalk (Houston 1)
ViaTalk (Houston 2)
ViaTalk (New Jersey)
ViaTalk (New York)
ViaTalk (Richmond)
ViaTalk (San Jose)
Virgina Broadband
Vozsip
VoiceEclipse
VoicePulse Connect
VoicePulse Open Access
VoIPTalk (a.k.a. TelAppliant)
VoIP User (voipuser.org)
Vonage (softphone account)
Voxee
WorldDialPoint (Australia)
If you are a service provider, support open device configurations, and would like to be included in this configuration wizard, or we are providing incorrect service configuration information, please contact us and we will do our best to correct the situation.
 
     
  Step 8 - Enter your Caller ID display name for PSTN Line VoIP Provider  
 
Enter the name you would like show up on outbound VoIP calls that originate from the PSTN-to-VoIP Gateway, assuming the function is supported by the VoIP provider. This name will also show up if an inbound PSTN calls rings through to Line 1 and the SPA3000/3102 is unable to decode the Caller ID information sent from the PSTN Line.
 
     
  Step 9 - Enter your User ID for PSTN Line VoIP Provider  
 
You'll need to enter the User ID, as given to you by your service provider, in the box below. Some providers may have you enter something else in this field.

Note: If you are using the same service provider on Line 1 and the PSTN Line VoIP configuration, either ensure your service provider supports multiple SIP registrations or use a different set of authentication credentials for the PSTN Line configuration.

iConnectHere Users: If you registered for a Receiving Calls service, please enter your personal phone number (i.e. 12125551234) here, and proceed to step 10. Otherwise, enter your iConnectHere UserID and proceed to step 11.

 
     
  Step 10 - Enter your Auth ID for PSTN Line VoIP Provider (not usually required)  
 
If you used the Auth ID field for something other than the SIP username/User ID, enter your User ID as assigned by your service provider. Otherwise, leave this blank.

iConnectHere Users: If you registered for a Receiving Calls service, please enter your UserID as assigned by iConnectHere. Otherwise, leave this field blank.

 
     
  Step 11 - Enter your password for PSTN Line VoIP Provider  
 
You'll need to enter your password as assigned by your service provider:
 
     
  Step 12 - Enter your Dial Plan for Outbound Calling from Line 1  
 
The Dial Plan tells your SPA what numbers are allowed to be dialed. It also allows you to do things like 7-digit dialing and dial out via different service providers you have configured in the Gateway Accounts section. In the Dial Plan field below, we've entered a generic Dial Plan that will work with the majority of providers. It allows you to dial:
  • 211, 311, 411, 511, 611, 711, and 911 are routed out the PSTN Line.
  • An arbitrary number of digits, which will be routed out the Line 1 VoIP Provider configuration;
  • dial * then dial one or more digits (routed out Line 1 VoIP Provider);
  • dial ** then dial one or more digits (routed out Line 1 VoIP Provider);
  • dial #1 then dial one or more digits (routed out Gateway 1 Provider, which will be the same a the PSTN Line provider in our Wizard);
  • dial #9 then dial one or more digits (routed out PSTN Line);
  • Dial #9* then two digits (routed out PSTN Line, i.e. for star codes).
 
     
  Step 13 - Select a Dial Plan for PSTN to VoIP Calls  
 
Used to restrict the numbers that can be dialed when calling in from the PSTN line into the VoIP Gateway. This will be stored in Dial Plan 8 on the PSTN List tab.

Note: It is recommended you leave this at the default value.

 
     
  Step 14 - Select a Dial Plan for VoIP to PSTN Calls  
 
Used to restrict the numbers that can be dialed when calling in from the VoIP line into the PSTN Gateway. Note this does not affect the dialplan used when Line 1 makes a PSTN call, only calls that enter via the VoIP Provider configured on the PSTN Line. This will be stored in Dial Plan 7 on the PSTN Line tab.

Note: It is recommended you leave this at the default value.

 
     
  Step 15 - Set Answer Delays and Ring Thru  
 
These options can be used to control how long it will take before the SPA3000/3102 answers VoIP to PSTN and PSTN to VoIP Gateway calls. You can also control if calls to the PSTN line will ring thru to the device attached to the Line 1 port on the SPA3000/3102 with an appropriate delay to capture PSTN Caller ID information.

The ring-thru delay is needed in order for the SPA3000/3102 to detect Caller ID, particularly with Bellcore-style Caller ID (used in the US). A minimum of 4 seconds is recommended to capture US-style Caller ID.

Note: The SPA3000/3102 does not support all types of Caller ID on the PSTN port. Bellcore (used in the US) is known to work.

Yes No
 
     
  Step 16 - Set Access PINs  
 
You must establish two different numeric PINs: one to allow access from the VoIP side to access the PSTN port, and another to access the VoIP port from the PSTN side.
 
     
  Step 17 - Set Your Timezone  
 
Select the timezone you wish your device to display the time for Caller ID in terms of. Note that these timezones do not take into account Daylight Savings Time.
No Time Zone
GMT -12
GMT -11
GMT -10
GMT -9
GMT -8
GMT -7
GMT -6
GMT -5
GMT -4
GMT -3.5
GMT -3
GMT -2
GMT -1
GMT
GMT +1
GMT +2
GMT +3
GMT +3.5
GMT +4
GMT +5
GMT +5.5
GMT +5.75
GMT +6
GMT +6.5
GMT +7
GMT +8
GMT +9
GMT +10
GMT +11
GMT +12
GMT +13
 
 
     
  Step 18 - Configure your adaptor  
 
Once you've entered the information above and checked to see if it is correct, click on the "Configure My SPA3000/SPA3102" button below. A new window will open allowing you to upload the configuration options directly to your adaptor. If you would like to re-enter the settings information, you may clear this form by clicking on the "Reset Form" button.

Note: Voxilla does not store any information that you enter on this form. Furthermore, we encrypt all submissions/responses to this form via SSL.