[humba3]

I have the setup from the sticky thread working.. if I dial 9 on my phone I get the PSTN dialtone, and I have a hotline cutover DP on my SPA3000 so that any incoming PSTN call is routed to one of my lines on the SPA9000.

So far so good. Now Linksys advertises the SPA3k series as being able to provide a backup line in case your Internet link goes down. Am I expecting too much if I'd expect this to work seamlessly by having one of my 4 lines on the SPA9k act as link to the SPA3k. Here's how I tried to get this working:

I configured my groups so that my Line4 is the last on each group.. so if the Internet link is down, it'll be the only line that still works.. so I have a seamless switch to PSTN.
I configured my Line4 to not register, and set the proxy to iport of my SPA3k, and the userid to the userid configured in the PSTN line of the SPA3k. I also set the line to make and accept calls without registration.
On the SPA3k I kept the config I had from the previous integration: same userid as in line4, no registration, and make and receive calls without registration.

If I cut the Internet link, all I get is the busy tone though. I haven't set up a trace yet (I will in time) but I'm wondering if anybody managed to get this working. I also tried adding a hotline cutover dp for line4.. still that's a nogo.

[mberlant]

The feature you read about is most likely the ability to configure your SPA3000 so that when it loses its internet connection it throws the power failure relay, which hard straps your PSTN line to the phone connected to Line 1 of the same SPA. If I understand correctly, this will not occur if the the SPA remains Registered with your SPA9000 when the SPA-9000's connection with your VoIP service provider goes down.

[humba3]

Here's the product page that mentions the feature: http://www.linksys.com/servlet/Satel...VisitorWrapper

If Internet service is lost to the SPA9000 or Internet service is down, calls can be redirected to a public telephone network using the SPA3000 Analog Gateway so service continues without interruption.

In my eyes that can only mean one thing.. it's a fully automatic process and requires no user intervention. And the only way I see this working from the SPA9k's perspective is having one line that goes to the PSTN gateway.. and since the SPA3k is no proxy, it also means that line must never register and must be configured to make and accept calls without registration.

Seamless switchover clearly is a business feature.. you can't expect business customers to remember to dial another prefix to go out via PSTN.. especially since there's no way of knowing (from your average user's perspective.. when you pick up the phone, dial and get a fast busy it can mean many things: incorrect DP, incorrect number, forbidden number, IPTSP has a problem, SPA9k has a problem or the Internet link is down) why you can't place your call.

[humba3]

I asked Sipura Support and that scenario is not possible.. so I asked if they plan to rewrite the product page to make it more clear there's no seamless switchover and that loss of connectivity to your IPTSP (be it a complete failure of your WAN uplink or just a problem on your IPTSP's end) always results in service interruption.. people will have to be told to use the PSTN line seizure prefix to dial out until the Internet link is back up.
Seamless switchover only works for the FXS port on the SPA3000 and since you can't connect that to the SPA9000, I'm SOL.

[humba3]

Hmm.. they came up with a solution after all. Here's what I had to change with respect to the original configuration:
Line 4 on my SPA9k uses a different userid than the PSTN line on the SPA3k. The proxy address on the PSTN line is set to iport of Line4 on my SPA4k, not ip:defaultport (so e.g. 192.168.1.4:5063 for line4 on the spa9k with ip 192.168.1.4).
On top of that, the VoIP to PSTN DP on the SPA3k is now: (xx.<:@gw0>) (from the normal Swiss DP with a 0 as outside line prefix).

Two problems remain: incoming calls cannot be put on hold/transfered (they are lost in nirvana), and incoming calls aren't cleared when the VoIP callee hangs up.. the PSTN Caller needs to hang up for the call to be cleared.