I’m trying to design the high level architecture of the solution for this client. They are a ~45 room hotel. They have an existing PBX solution, with traditional analog 2-wire phones to every room, hooked into this PBX system (unknown), with terminates/joins the phone system through a PRI or BRI ISDN (likely, but don’t know for sure).
Goal: replace this aging system with a linux/asterisk/voip solution.
(1) the client would likely need to rewire all rooms for VOIP/ethernet capable handsets. Not a problem, labor is cheap, and while rewiring we can use Cat6E which will allow us to use 2 pair for voip/sip handsets/phones and the other two pair for ethernet for the room.
(2) The client has only a DSL line. We’re going to try to get them a 512kbps SDSL or even a full T1 to handle the anticipated call volume. Worst case is 82kbps (64kbps including IP overhead) for 20 rooms concurrently. I’d like to see if we could tweak that down to ~64kbps (including overhead) for each call, but who knows. I’d like to know what people have gone down to and still experienced good results.
(3) The hotel is in a central american country. Most of the folks in the rooms will be calling either Europe (15%), US (70%) or other parts of the world and/or LATAM (15%).
Although I can do an entire solution based on Linux/Asterisk/VOIP (LAV), it has to terminate into a VOIP->PSTN gateway somewhere. This is what I’m having trouble understanding. All hotel rooms, and all guests will be dialing traditional phone numbers as their destination. Thus, VOIP only gets us to the other VOIP ‘side’ (as in provider – I don’t know what the term for this in VOIP-verse is called, but in PSTN it’s the destination CO).
Who provides termination or VOIP->PSTN/POTS GATEWAY services. The hotel, again, is central america. The provider would likely be in the US, no?
How do I determine rates? What is the reasonable going rate?
Thanks for helping me in my first design.