Sip2Sip > voicemail / mobile - is this a "safe" config?

Let's say I have an Asterisk server on Voipfone extension 200.
My home Gigaset VOIP phone is on 201.

Say I purchase extension 202 and attach a SIP device to it, then set up forwards like this:

Try 201. When busy/no answer, go to voipfone voicemail. PSTN failover = my home "real" phone number.
(I'm assuming here that PSTN failover means dial a "real" number when VOIP fails?)

Reason? There's a "web" dialler which uses sipml5, meaning you can call directly from a webpage in Chrome or Firefox, but it must be a TLS connection which voipfone doesn't have. So, I create 2 sip2sip accounts - one attaches to Chrome securely, the other attaches to Voipfone non-securely, so it's:

Chrome > sip2sip1 > sip2sip2 > voipfone.

That way:

1: My passwords are secure
2: Know-one knows where the call is ultimately ending up.

Is there ANY way someone can override the diversions I have setup and make phone calls to Ghana or something?

In other words, am I inadvertently opening up a gateway of doom?

Does any of this make sense?!? By the way, it's fine "in theory" as I can call a Microsip softphone using the above Chrome > sip2sip1 > sip2sip2 > microsip , but of course, that doesn't have the ability to dial out to the PSTN like Voipfone does.

I know I'm being "cheap" not using Voipfone's own perfectly good web button, but I thought I'd try it as an experiment :)

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