SPA3102

This is a page about a SIP ATA adapter, I think I still have one.

What it can do

It has 1 FXO and 1 FXS port. That means that it can connect to a PSTN line (The Phone Company) and it can connect to (one or more) analog handsets.

It has two ethernet ports, one for your internal network and one that is intended to connect to the Internet.

PSTN fallback mode

If the SIP proxy fails, it can connect the handset(s) on Line 1 directly to the PSTN line.

Router features

It supports a few basic router features including port forwarding and DMZ passthrough. It can be a DHCP server. It supports QoS and VLAN settings.

Streaming audio server (SAS) feature

When enabled, incoming calls are auto-answered and a stream of packets goes to the caller. So the line could be used for something like listening to the output of a microphone or a radio or a recorded message.

Hooking it up

Serial number: FM600GA18536
Hardware version: 1.4.5(a)
Software version: 5.1.10(GW) (upgraded from 5.1.7(GW) on 24-Oct-2009)
Mac address: 000E08CEB151

I hooked it up and put a static IP address on the network side. I turned off the DHCP server.

I plugged a phone into the phone jack and configured Line 1 as I would any of our other Linksys/Sipura phones (PAP2, SPA841, SPA941).

I added entries for the phone in my Asterisk extensions.conf and sip.conf files. I reloaded asterisk and then checked for registration of the phone.

asterisk -r
CLI> core set verbose 10
CLI> reload
CLI> sip show peers

It shows there with an ip address of 127.0.0.1 which can't be good. :-)

Solution: You must use the INTERNET port, not the ETHERNET port! REMEMBER also that you must enable the Web GUI on the INTERNET PORT! Don't do what I did!

Here is some more help: http://blog.pathennessy.org/2009/01/01/configuring-linksys-spa-3102-for-asterisk/comment-page-1/

Another approach is to use this configuration wizard, which directly reprograms your SPA3102 and also tells you the settings to add to your Asterisk server.

http://voxilla.com/voxilla/tools/device-configuration-wizard/linksys-spa3102-spa3000-configuration-wizard-for-asterisk

Outside person can't hear me!

If people complain they can't hear you, turn up the Input Gain. It's on the Regional settings near the bottom of the page. Default is -3, I set it to 6 for our Western Electric phones. Then I called the Twilio test number to test it.

Links

Copies of the manuals:

SPA 3102 administrative guide http://homepage.ntlworld.com/paul.steel/voip/