My project is to enable people in 2 different countries (UK & Spain’s Ibiza) to communicate easily & cheaply with each other using their mobile phones. IE their mobile phones would be used as extension devices on the asterisk servers.
How I envisage the work/call flow:
---------------- UK ---------------- /////---------------------- Ibiza --------------------------
mobile -> gsm gateway -> asterisk server in UK -> Asterisk server in Ibiza -> pstn gateway -> mobile
In my experience, a PSTN interface is always outgoing, not incoming. If all the people here use mobiles instead of IP phones, I assume I need enough gsm gateway channels to handle all concurrent incoming calls.
Am I correct regarding the call flow? I cannot find any mention of gsm gateways on Digium’s website/equipment list, although plenty of gsm gateways visible on google.
Can a PSTN interface handle incoming gsm calls and interface to asterisk (or another ip pbx)?
IE can I just buy in a PSTN interface & get them to handle the gsm gateway stuff?
I understand from a little googling that SIP trunks replace local PSTN interfaces, although I’m not clear if they would be relevent to my purposes here, since my use of voip here is to bridge the gsm systems in 2 different countries, therefore I suspect the need for local PSTN interfaces in each country.
I don’t believe that is legal in the UK. You cannot resell capacity on fixed GSM terminals. You can only use them for traffic to, from, or within your own business. Similar restriction exist in other countries, quite possibly in Spain.
Asterisk General is for discussions about Asterisk, not for obtaining advice.
I don’t understand what you mean by a PSTN interface always being outgoing. That’s not true in any sense that I understand.
Maybe my post was too vague in its wording; my project is intended for internal “group” comms, not for reselling.
I posted here as a place to start to look for solutions for this voip idea, especially since my whole call routing would involve using & interfacing to asterisk servers.
My only exposure to PSTN interfaces was using a voip gateway with an IP phone. I apologise for the waste of space, I used that gateway mainly for outgoing calls, forgot that I had an incoming number assigned to me.
Legality issues aside, yes you can do this easily. Digium themselves do not sell a GSM gateway, but there are several options for that. You can either get an Asterisk-compatible PCI card that fits in the server with 1-4 GSM modules/SIMs on it or you can get an external GSM gateway that speaks SIP to Asterisk.
Once you get the call into Asterisk via either the PCI card or via a SIP gateway, then you can route the call anywhere you want to (a voip provider, another Asterisk box over the internet, or back out another GSM channel). So your general call flow is correct, you’ll just need to work out the details of how you want the calls to work, such as “When I dial the number associated with one of the GSM gateways, does it directly dial someone else or do I get a second dialtone I can use to dial where I need to go.”