How does it work??? - PBX Info :: Your Free PBX, PABX and Telephone Information Resource

WhoURDaddy · 04-02-2005, 12:13 AM

I'm new to the VOIP world and need some advice. I currently have a
Panasonic KX-TD-816 PBX installed in my central office... and have some extra unused telco lines available on the PBX. I'd like to use one of these extra lines for VOIP into the PBX/ phone arrangement. I bought a Digium wildcard x100p card for use in a spare computer we have in the office to experiment with Asterisk... but still don't know much about all this.

Question: Is there a good way to patch the VOIP/internet into my Panasonic PBX using a computer and a card like the x100P or some other FSX? Is there some basic text somewhere that spells this all out?

Why do I want to do this? Well all of out sattelite offices and warehouses have KX-TDxxxx equipment and space telco lines too... so if I can get the main office working.....

You can reply to bill@wjschmidt.com

**rixride** · 04-27-2005, 08:11 AM

HEy Bill
Have your read how to build a pbx

The Asterisk pbx isn't a VOIP service, you will need a VOIP Service such as Broadwing or Vonage to send your calls over the internet.

The digium line card will allow you to use your Asterisk PBX and Sip phones and make calls through the analog line that you have attached.

The only VOIP calls you would be making would be a SIP to SIP call if you have 2 or more PC's with SIP phones on them. But when you try to make an outgoing call it would go over your analog line.

Does that make sense?

**rixride** · 04-27-2005, 08:42 AM

Here is a image of what I am talking about.

After set up the Asterisk Server, You can connect it to your internal network. Personally I have a Linksys ethernet switch. All my computers reside on that switch. I have also downloaded SIP phones for each computer.

So SIP User A can call SIP User B and that is a VOIP Call traversed through our internal network.

If SIP User A does not have a VOIP Provider (i.e Vonage) when calling an outside number like an 800# it would go out PATH A . Using the Digium card and the analog line.

If SIP User A does have a VOIP Provider then when they go to make a external call it would go out PATH B using VoIP most of the way.

I hope that helps.

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