Here’s a full guide on how to setup a Linksys SPA-3102 VoIP ATA to use a VoIP Provider for ADSL Connections, step by step.
Place the SPA-3102 in a place that’s convienient for you, generally next to your ADSL modem is ideal.
Here is how everything connects, looking at it from a view of Phone socket, down:
The phone socket connects to a double adapter.
The double adapter splits the phone line into two, one for ADSL, the other for your SPA-3102 ATA.
Plug the lead from the ADSL modem into one port of the double adapter.
Plug another lead from the other port of the double adapter into an ADSL filter.
Plug a network cable into the Internet Port of the SPA-3102 ATA into the Modem / Router.
Note, if you have a single port ADSL modem, unplug your computer, and plug it into the Ethernet port of the SPA-3102.
Plug a regular phone handset into the Phone port of the SPA-3102.
Finally, power the device on.
Now, the device needs to be configured.
Pick up your handset and dial ****.
Dial 110#, and listen for a series of numbers, write this detail down (or get yourself a notepad window).
Place the phone back on its hook.
Plug a lead from the ADSL line filter into the Line port of the SPA-3102.Watch movie online The Transporter Refueled (2015)
Next, open your web browser. In the address bar, type http://, followed by the IP address of the SPA-3102, followed by “/advanced”. eg. http://192.168.3.60/advanced.
On the Router Tab, click Wan Setup, and choose DHCP from the connection types list.
On the Router Tab, click Lan Setup, and choose Bridge from the connection types list (important for those who only have a 1 port modem / router, not a bridged modem however).
In the address bar, type (using example IP address above): http://192.168.3.60/admin/voice/advanced.
You can do exactly the same by clicking the Voice Tab, choosing Admin Login, choosing Advanced.
Click the Regional Tab, and enter the following settings in Call Progress, for each of the fields:
Dial tone: 400@-19,425@-19,450@-19;10(*/0/1+2+3)
Busy Tone: 425@-19;10(.375/.375/1)
Reorder Tone: 425@-19,425@-29;60(.375/.375/1,.375/.375/2)
Ring Back Tone: 400@-19,425@-19,450@-19;*(.4/.2/1+2+3,.4/2/1+2+3)
MWI Dial Tone: 400@-19,425@-19,450@-19;2(.1/.1/1+2);10(*/0/1+2)
Scroll down some more, to find Distinctive Ring Patterns, and edit as below:
Ring1 Cadence: 60(.4/.2,.4/2)
Scroll down some more, find Miscellaneous, and edit:
FXS Port Impedence: 220+820||115nF or 220+820||120nF
DTMF Playback Length: .25
Time Zone: GMT+11:00
Click “PSTN Line” near the top of the page.
Find PSTN Disconnection Detection, and edit:
Disconnect Tone: 425@-30,425@-30;1(.375/.375/1+2)
Find International Control, and edit:
FXO Port Impedance: 220+820||120nF
PSTN to SPA Gain: 3
On-Hook Speed: 26ms (Australia)
Click the Line 1 Tab, near the top of the page.
Scroll to Proxy and Registration, and edit (your settings replaced from your provider):
Proxy: Enter Proxy Server Address
Outbound Proxy: Enter Proxy Server Address
Use Outbound Proxy: Yes (if your provider supports it)
Register Expires: 1800 (30 minutes is better, ensures you are always registered to receive calls, the lower, the better).
Use DNS SRV: no (if your provider supports it, use it).
Proxy Fallback Intvl: 1800 (see Register Expires).
Use OB Proxy in Dialog: yes
.. The other settings should be fine as “No”, but if your provider supports it, “yes” is good.
Scroll to Subscriber Information, and edit:
Display Name: Your Name (this name can appear on Caller ID for SIP to SIP calling on the same network as well (in some cases)).
User ID: User ID (your user ID)
Password: passw0rd (your password)
Use Auth ID: yes (most providers support Auth ID, those that don’t should set to no).
Auth ID: User ID (its nearly always user ID).
Scroll to Audio Configuration, edit:
Preferred Codec: G729
DTMF Tx Method: AVT, if dial tone systems, such as Push 1 for X, don’t work, change to AVT+INFO.
Hook Flash Tx Method: AVT
Scroll to Dial Plan, edit:
Dial Plan, depending where you are located, and what calls you want to send anywhere, you’ll edit this to:
(*xx.|000S0@< :gw0>|< :02>xxxxxxx|0xx.|1x.< :@gw0>|< **1:>xx.< :@gw0>)
This dial plan allows NSW same state dialling, dialling other states and mobiles, sends 13, 1300, 1800 numbers to your land line, and allows prefixing calls with **1 to go through PSTN line, sends 000 to the PSTN, and allows use of * feature codes.
Scroll to the end of the page, click the “Submit All Changes” button, and enjoy the ATA, which should now be perfectly operational.
Pingback: Adventures in VOIP (Part 1) - Two plus Three