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Linksys SPA2102 and SPA8000

MarkW S7

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Going ahead and making a thread for these devices. If anyone has any questions regarding these let me know and I'd be more than happy to share.


Linksys SPA8000

This is an 8 fxs port ATA (Analog Terminal Adapter or Analog Telephone Adapter). You can either connect the lines through the fxs ports or through a RJ-21 (50-pin telco connector). That is great if you want to maintain the carrier demarc style cross connect. Beyond the physical connections, internally the spa8000 offers a whole host of great features and flexible voip settings including offering 4 internal trunks. One huge advantage that I feel that this and the spa2102 have over other ATA (like the Budget Tone or Audio Codes MP's) is that each configurable line is proxy independent. What I mean by this is Line 1 can talk to a completely different proxy or outbound proxy from Line 2. Think of the possibilities in redundancy carriers, a bridge between 2 domains, etc.. The only major thing to note about these devices is that they can only run in a LAN/Host mode, and not as the gateway/router like it's brother the SPA2102



Linksys SPA 2102

This device uses the same OS that the SPA8000 uses and has almost the exact same functionality and features. The main difference is that the 2102 can run as a basic router with dhcp, nat, and limited QoS. Other differences include: 2 fxs ports, device can be set up as a switch and not a router, no trunking features.


Both devices support t.38, adjustable DTMF mode/methods, a whole slew of settings that would clutter this board very quickly.

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  • 7 months later...
Guest Fred

Hi! Mark,


I was trying to auto-provision my LinkSys SPA2102 from scratch to PBXnSIP PBX, can you please send me the procedures to create the provisioning from scratch?




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  • 1 year later...

Hey Mark,

we are using an asterisk v1.8 system with snom 320 phones. After having registered, without aktiv using the spa 2102, the snom320 crashes down after 1 or 2 days.

Trying to start a phonecall, the phone rings and shuts asterisk down after the connection is established.

Any ideas?

what is an theoreticall way to disrupt the snom via sip?

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