So, I was buying a new phone, and that phone happened to only have 1 sim card slot, the other sim is an esim (which isn’t supported yet where I live), while I have dual sims that I need with me at all times. And so I thought, is there a way to host a server on my old phone (or a computer/raspberry pi/linux server) connected on the same network as that old phone, and have an app on the new one that allows me to dial a number, click on call, and in the background the server handles the request by initiating the call on the old phone, and directing mic and sound to the new one? The phones and potential server could be connected to a VPN tunnel of sorts so that the ips of all 3 devices would be static and known at all times.

It could be done with some programming, which I could do, but I’m admittedly too lazy, so I thought maybe there’d be a way to do it without? Maybe an app that already does this or anything similar? A google search didn’t return anything useful, but maybe someone here knows something.

  • NikStalwart@alien.topBEnglish
    1·
    1 year ago

    You are looking for VOIP.

    Asterisk is the main goto, but it is still waiting for patches to some RCE vulns disclosed last DEFCON.

    Your country might not support using a mobile number with VOIP providers, mine does though.

    Good luck.

  • Tekrion@alien.topBEnglish
    1·
    1 year ago

    I don’t think there’s a ready-made solution for something like this. If you were going with the coding route and we’re just concerned about making calls and not receiving them on your main phone, then you could verify your second number with a SIP carrier like twilio so that you can use it as an outgoing caller ID with them, create a sip trunk with them, spin up a local asterisk server and hook it up to the sip trunk, and then program your asterisk dialplan to allow you to dial out using that second number as the caller ID. Then you’d just use an app like linphone, sign in using your asterisk credentials, and dial out from there. It’s a lot of setup to do if you’re not familiar with asterisk or sip trunking though