I spent a lot of time tracing and talking to Sipura support so I figured I'd share my findings.
First of all, if you want to use your SPA3k as an analog gateway for the SPA9k, kiss the hop-on, hop-off functionality goodbye - you can't have both.
Once you've come over the shock of that, there are two ways to achieve interoperability, the IP dialling and the proxy way.
IP dialling means you don't need to sacrify any of the 4 lines on your SPA9k for the interop - in return you don't get seamless switchover to PSTN if your Internet link should go down (at least not for all the devices connected to your SPA9k.. it will still work for the analog phone connected to the FXS port of the SPA3k), and your devices need to support IP dialling or dialplans with substitution sequences (all Linksys SPA devices support both - most other devices should support at least IP dialling). Another drawback is that certain transfer scenarios won't work: VoIP user calling out via PSTN, then transferring the call to another VoIP user - the IP end of the call will be dropped while the PSTN call will remain active indefinitely (unless you have VoIP silence detection).
If you go the proxy route, you will need to use one of the 4 lines on the SPA9k, and in order to support direct selection of the PSTN trunk, you need to set up
trunk seizure prefixes. The upside is that in case your Internet link goes down, calls can automatically be routed to the Line on the SPA9k that communicates with the SPA3k, and transfering calls is not an issue either.
Now that I have outlined pros and cons, let's have a look at the setup. The SPA3k side will remain the same regardless of which interop mode you chose.
First of all, disable all NAT settings for the PSTN line.. you won't be needing them. Then set the proxy to
<ip-spa9k>:<port line x>. The port is either 5060 (line 1 on the SPA9k), 5061 (line 2 on the SPA9k), 5062 (line3) or 5063 (line4). Turn off the use of the outbound proxy (Use Outbound Proxy & Use OB Proxy in Dialog), as well as Register, and activate "Make Call without Reg and Ans Call Without Reg".
Why these ports you'll ask? The SPA9k only accepts SIP messages on 6060 port from devices that are registered with the SPA9k.. since the SPA3k isn't registering, it can't send messages to that port and needs to use a port where external servers may send messages to.. and those are the ports that are used for the 4 lines on the SPA9k. Now you'll wonder why not simply register the PSTN line with the SPA9k.. we'll get to that later.
Either way, in my case, the SPA9k has an IP 192.168.1.4 and I'm using Line 2 so the proxy becomes 192.168.1.4:5061.
In the Subscriber Information section, set the Display Name to whatever you feel like, and the User ID to a free extension in your SPA9k dialplan (e.g. I am using 2 digit internal numbers and 89 was unused so that was my User ID). No password, or Auth ID is required.
Then the dialplans. You'll need one for incoming calls and one for outgoing.
I set the outgoing DP to (xx.<:@gw0>) (any 2+ digit numbers will be sent out via PSTN), and the incoming DP to
<:88@192.168.1.4:5061>
What the heck is that all about? This is a hotline cutover DP. No matter who calls from the PSTN line, the call will immediately be forwarded to 88@192.168.1.4:5061. You may recall that 192.168.1.4:5061 refers to line 2 on my SPA9k. 88 is the user ID configured on line 2 on the SPA9k. If your user ID on that line is 12345678, then the DP would become <:12345678@192.168.1.4:5061>
Then turn on the VoIP-To-PSTN Gateway Enable, set VoIP Caller Auth Method to none, Make sure One Stage Dialing is turned on, and set the VoIP Caller Default DP to the outgoing DP (xx.<:@gw0>). Note that this is a very simply DP, you could of course come up with more complex stuff, but since this is an analog line, every digit you dial is sent out immediately, and your PSTN provider will play a prompt if you dial something that cannot be dialled, so why bother replicating your regular DP here.
Also turn on PSTN-To-VoIP Gateway Enable, set PSTN Caller Auth Method to None, and if you want the PSTN Caller ID to be forwarded to VoIP, turn on PSTN CID For VoIP CID. Also set the PSTN Caller Default DP to your hotline cutover DP.
Last but not least, besides getting the International Control settings right, the PSTN Disconnect Detection settings are rather important. Sipura suggests that you do not use Long Silence detection.. it's really a hack and if somebody puts you on hold without actually using hold (so just puts the phone down and goes someplace), you may lose your call unless there's background noise. Whether you should turn on CPC or polarity reversal depends on the country you live in and I can't make any suggestions. The Disconnect tone is extermely useful.. it's the most reliable way to quickly recognize a disconnect so your FXO port will be available again once the call has ended. Only if you can't get it right and if CPC and Polarity Reversal don't help either should you resort to silence detection.
Alright, now we're done with the gateway.
Let's look at
IP Dialling first:
Your Phone DP will need a substitution element that directly dials the PSTN line on your SPA3k:
<9:89@192.168.1.3:5061>S0
This one immediately gets you the PSTN dialtone on the SPA3k once a 9 has been dialled. Since 9 is often used as an outside line prefix, you might replace it with something else, perhaps #9. The rest will appear familiar: 89 is the user ID we've used for the PSTN line on the SPA3k, 192.168.1.3 is the IP address of the SPA3k, and 5061 is the SIP port of the SPA3k's PSTN line. On top of that, make sure your phone is configured to make calls without registration (the phone isn't registered to the SPA3k, so unless
Make Call Without Reg: is activated, dialling 9 will fail).
A device that cannot have a substitution DP would simply have to dial 89@192.168.1.3:5061 to get the PSTN dialtone.
And now the
proxy method:
You first need to pick one of the 4 lines of your SPA9k to do this. Then you configure it very similarly to the SPA3k: No NAT settings (NAT Mapping Enable and NAT Keep Alive Enable), the proxy is <ip-spa3k>:5061 (5061 is the default SIP port for the PSTN line VoIP account), no outbound proxy, Register: No, Make Call Without Reg & Ans Call Without Reg: yes.
If you want, you could reproduce your regular DP on either the Line X DP or on the phones. I decided not to do that but if you want to restrict diallable numbers, it's something you should do and since it's simpler to just configure it on the Line X, that's where I'd do it.
You can pick anything you want as display name, and user another free number within your numbering plan (I picked 88) - once again there's no password, and no auth ID, and I set the call capacity to 1.. there's only one call via PSTN after all. Don't forget to configure your contact list accordingly.
Now, for the seamless interoperability, the call routing rules come into play. By default you have one rule: every call is routed to line1, if it's not up, line2, then line3 then line4. If you want to keep that, use Line 4 for the interop.. that means that if all sip accounts fail, the PSTN line will be used.
If you are using line groups instead, configure it as you want. If you are familiar with DID number based call routing, you could also set up certain calls to directly go to the PSTN (e.g. 911) - either via call routing rules (the prefered way imho), or via ip dialling in the phone DP (see the IP dialling solution for details).
And that's all there is to be said about this.
Since you have to give up hop-on, hop-off on your SPA3k to do this, perhaps you'd then like to use the FXS port on your SPA3k as another line on the SPA9k.. if that's the case, just set it up to register with your SPA9k, just like you set up a regular phone. Or, if you want to keep the external SIP line, you could just set up an IP dialling DP element that allows the analog phone to call extensions on your SPA9k:
[1-9]x<:@gw1>
That element routes all 2 digit numbers to gw1, which is 88@192.168.1.4:5061 (in case you forgot, that's the same URI as used on the SPA3k for the hotline cutover).
To enable SPA9k extensions to call the SPA3k Line1 phone, you basically have the same two options as outlined above: either a DP element, or using yet another SPA9k line for the interop (I doubt many people would want to do this). For the IP dialling solution, if the user ID on your SPA3k Line 1 is 31312345678, then the substition element would be <#1:3131234567@192.168.1.3:5060> - and of course the Line1 on your SPA3k would also have to be configured to take calls without registration.
Last but not least, why not register the SPA3k with the SPA9k? I've tried back in the day, but then my hotline cutover DP didn't work anymore.. the problem is once you are on the "inside" of the SPA9k, you can't make calls to the outside (e.g. userid-line2@ip-spa9k:5061) anymore, so your hotline cutover would have to go to an internal SPA9k extension.. while aa@ip-spa9k:6060 should work, if you don't want to use aa, the only alternative is route the call to an extension directly.. if that works for you, fine - I want it on a line so I can use the normal call routing rules with all the bells and whistles.
I suppose this will raise some questions for some, so here's my plea: before you ask, make sure you read the entire instructions about 3 times and try to configure each element in the order outlined.. it's not a simple procedure and it's not easy to wrap your brains around it (all this is info gathered during a two month period and I spent an excessive amount of tinkering and tracing in the process). If something doesn't work.. make sure you provide traces from the getgo (SPA3k PSTN line and SPA9k proxy).
SPA3k & SPA9k interop explained 
Linear Mode
