NEWBIE to the telephony world


#1

Ok… when it comes to computers and networking, I am very fluent, so feel free to lay it on me. This may sound like a stupid question, but I’m going to ask anyway.

In setting up the initial test project… this is what I’m going to use and am just wondering if it will work. I am a little shaky on it because I’m not that familiar with VoIP, not yet.

1.) Linux machine running asterisk
1.) VoIP line directly from att
1.) 10/100Mbit switch
3.) SIP phones

Now I’m going to keep this totally seperate from data, so it will only be used for voice. In setting up this configuration, can i just use 100MB ethernet cards in the linux machine? So basically it will look like this…

VoIP-----------------eth0=[ PBX ]=eth1---------------[12 port switch]-----------phone1,2,3

where ---- is the cabling

Maybe I should be looking at the dlink router that is currently there for im not even positive if the phone line coming in is RJ45 or 11… But anyway, will I be able to configure asterisk to use the ethernet cards, and basically router the phone call coming in from eth0, to eth1 to its desired port(extension)? Like, can asterisk configure which port on the switch should be the extension? So if I have 12 ports total, plus the uplink connected to eth1, can i setup ports 1-12 to be like, ext. 101, 102, 103, etc etc. And also, will this setup work using regular ethernet cards… Now I did purchase additional X100P cards, but that’s going to be a different portion of the project after I get this initial setup working. Any information will be highly appreciated, for all of my google search topics are still leaving me wondering… Thanks.


#2

Ok, I totally take back the phone call routing portion. I completely overlooked the IP addressing of the phones and assume that would be in a config file for asterisk. Sorry. So my question I suppose is just the ethernet setup…


#3

Yes, just normal ethernet cards.

Although i don’t understand the [quote]phone line coming in is RJ45 or 11[/quote] bit. You sound like you’re talking about a normal POTS phone line and plugging it into an ethernet card. That doesn’t make sense. You also refer to a [quote]VoIP line directly from att[/quote] - what is that exactly? Are you talking about an internet phone service, or something different?


#4

Well, I’ve got two phone numbers, through at&t which are VoIP. The lines are split over a telephone wire (1 pair in each connector) and plug into my VoIP router and so on. I just checked it out, as I’m home from work now, and it is an RJ11 connector. So i guess I’ll have to convert the connector, although I’m not sure where the pair of wires should lay in the RJ45 connector…


#5

To be honest I thought the line coming in was already running on ethernet… I suppose I may have to wait for those FXO cards to arrive…


#6

Re RJ11/RJ45, have a look at an earlier post (always a good idea before you ask questions, anyway!). I found it through the search link on this forum - you could too…

forums.digium.com/viewtopic.php?t=471

And why don’t you ask AT&T what form of connections their service uses? I’ve never heard of VOIP over a phone line. If it’s sold as VOIP, i wouldn’t expect it to be POTS.


#7

Haha, yeah… I am being a complete idiot. I didn’t think it was over an analog line but the dlink routers we’re using are confusing me. I’ve got two numbers activated over a full T1 line. The routers ATT supplied us with are limiting is features and functions though i think. I setup asterisk and got the pbx functioning inside the LAN, but I think I’ve got more reading to do for the outgoing calls part. Given these phone numbers are active over the line, assigning the asterisk NIC an internet IP should make it ready to place outside calls no?


#8

I still don’t follow you really!

It depends what the NIC’s connected to. The internet? Setting an IP address on a NIC doesn’t achieve anything in itself - the NIC has to have a connection to something and the IP address you’re using has to be routeable from that “something”. And, to make and receive telephone calls, there has to be a VOIP termination service that you’ve got an account with somewhere in/on that “something”…

That “something” could be the internet, a LAN, or some other type of IP network.


#9

Sorry for being so vague. I’m really flaky on setting up the incoming/outgoing call portion of asterisk. When i said assign it an internet ip I meant from one of my Ip addresses. I’ve got a small portion of a C class, 32 addresses, 2 of which are being used now for my VoIP router. I’m still looking around for more documentation on this so i don’t have to post these stupid questions… :smile: Although, I dont think it’s going to work anyway though, being the service is ATT CallVantage, and I’ve already bugged their tech support for users/passes/proxies with no luck of getting them and have read other posts asking about setting it up using callvantage.