Can I build a forwarding system with a home computer and Ast

Hello everyone, ok I have been doing a bit of homework and it appears that Asterisk may be what I am looking for.

I am looking to provide a call forwarding service, it should be fairly simple coding (I suppose). Where I provide each subscriber with a local number that is then forwarded to a long distance number of their choosing. This service will have atleast 20 people using it at any given moment and each will have their own private number that is then forwarded…

Now, from what I understand I will need a T1 connection so I think I have found a provider for that. I also (believe) I have found where I can get the phone numbers for a reasonable rate.

Now, my question is can I build a forwarding system with a home computer and Asterisk? Again, I stress it will be simply when they call their number assigned to them I want this software to forward their number to a long distance number…

Thanks for any help you can provide.


Sure Asterisk can do that. Remember, if you are planning to have 20 concurrent calls, each consuming 2 channels, you are going to need at a minimum 2 T1’s to cover the 40 offhook ports.
Alternatively, you could put your server in a colocation and do everything over SIP.

Dave thank you for responding, what do you suggest the minimum requirements would need to be for the home computer (server)?

Also, I recall SIP having something to do with software phones… could you explain the alternative to 2 T1’s a little more? Thank you for your patience I am sure this is basics 101…

I suppose I am wondering if I can accomplish this with 1 T1 connection what I am looking at is a (1.5/1.5) bandwidth.

Again thanks

You cannot accomplish this with a single T1. Any current PC should be able to support that many concurrent calls just passing the audio from one side to the other.

The alternatives to 2 T1’s is putting your server in a colocation. These are places that allow you to put your server in one of their racks for a monthly fee. Included in the fee is a certain amount of bandwidth. With this option you would need to select an incoming DID SIP provider for your subscribers, and maybe a few different outbound SIP providers for fail over. There are many choices of sip providers and the general rule of thumb is you get what you pay for with them. Alternatively you could rent servers from a colocation provider and install asterisk on them. I’ve used serverbeach for this with good results.

With 1.5Mb connection you can get roughly 20 concurrent ulaw sip calls (Maybe less I didn’t use the bw calculator), each ulaw channel takes 80k). More if you use a compressed codec.

This was very informative… and greatly appreciated.