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Connecting a standard PBX to a VoIP gateway for inter-office calls.

In each of our three offices we currently have installed a standard digital Nortel Norstar PBX. Each PBX has 8 to 12 analog FXS lines to the PSTN and roughly 30 phones.
We would like to be able to make inta-office phone calls over the internet to reduce our LD costs.
My idea is to connect some of these FXS lines to a VoIP gateway so that we can call the other office by dialing some kind of prefix (like *21 for location 1 or *22 for location 2) and then the phone extension in that location.
I am comfortable on the IP side of this solution but not as much on the PBX side so I am trying to understand what are the things I should look for.
- can I still use these analog lines on the PBX or I should go with a T1 interface?
- How do you actually configure a Norstar PBX. Is this something I would be able to do myself or I should use some outside vendor?
- I can't find any Norstar user guides that easily explain how to configure these systems.
- numbering plan. is*xx a good way to identify the remote offices or I should do it some other way?

Any suggestions? Any good Web site to look at?

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1 Solution
I cant help with the Nortel configuration. How many calls do you want to support between offices?

For example what you could do ig get a couple of Sipura SPA-2000 devies for two sites and these will link to two FXS interfaces each. Then at the 3rd site get a box running Linux and install Asterisk on it and fit a couple of FXO interfaces (or a 4 port card if you wish). The asterisk box will then communicate over its own FXO interfaces and across the internet to the SPA-2000's.

The way you are planning to do the numbering sounds fine but if each office is using a different number range for their extensions you should be able to program the system so that you just need to dial the extension number.
excelacomAuthor Commented:
I think 3, maybe 4 simultaneous calls.
I am now evaluating two SPA-3000 here in the office. great products with a ton of features unfortunately only allow one call at a time.
Do you have any suggestion for a 5-6 lines gateway?

Your numbering plan is a good suggestion. we could do 2xx for the HQ and 3xx for office 1 and so forth, my question is how does the routing works if I am dialing 311 from the HQ, the PBX recognizes the first "3" as being office 1 so I believe it would start an outside call to a specific FXO which is the one where the gateway is connected. then how would it communicate the rest of the extension to the other PBX? is it DTMF? would only the "11" be sent to the other PBX?
Or does it uses some signaling like SIP?
I would really like to keep it as simple as possible.

2 lines - SPA-2000
4 lines - either two SPA-2000's or a pc running asterisk together with a Digium 4 line analogue card

You can keep adding SPA-2000's as long as you want but after a point it becomes expensive.
I only recomend installing a maximum of 2 telephony cards in an asterisk server due to the large amounts of interrupts produced. You can therefore effectivly go up to 8 lines before you need to switch to a T1/E1 interface.

Dialing does use DTMF over the analogue lines so if a connection between offices goes over two analogue interfaces this will add an amount of time it takes for a call to be established. You can often use the '#' symbol to indicate the end of a dialed number so if you can include this in your dialplan you could cut the delay.

Typically in a PBX you can configure a set of rules for sending calls out particular interfaces. Currently you probably have a rule which sais if the number starts with a 9 (outside line) then strip the first digit and send the call out of a group consisting of a bunch of analogue lines. You just need to do something similar with the new prefixes. I would not strip the initial digit as then call routing will be decided by the asterisk box.

You would have a Linux machine running Asterisk (Its what I use in our company) that would connect to all the SPA-2000's over the internet and also the local telephone system using a built in 4 port telephone interface. It will receive the calls and strip leading digits and pass it onto the different PBX's as apropiate.

If you want more than 8 lines then you should look at a E1/T1 interface installed in an Asterisk box on each site. This will push up the cost considerably though. It will give you more options though such as the ability to have people using VoIP phones and being able to use them as normal phones whichever location they are currently in (or even hotels etc...)
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excelacomAuthor Commented:
grblades, Thanks for all this good information.
One more thing, why should I use an Asterisk system when I could buy a dedicated gateway for $800 http://www.voipsupply.com/product_info.php?&products_id=1637.
Even if I decided to replace the Nortel PBX with an Asterisk system in the future,  I believe I would still be able to use this gateway "the other way around" to provide connectivity to the PSTN and save the money of the analog card for Asterisk. Does this make sense?

Last thing, From your comments I believe you have a full blown IP PBX system in your company, Do you necessarily have to run your VoIP telephones on a separate VLAN or you can just enable QoS on the switches and give higher priority to the VoIP traffic?


That device is purely an FXO interface so is suitable for connecting to the PSTN and FXS interfaces on a PBX (the same type of interface an analogue telephone connects to).
The SPA-2000 is similar as it is a 2 port FXO only.
This should not cause you any problems as your exchange will typically have lots of spare FXS ports on it. The only ports you cannot use on the PBX are its own FXO ports which connext to the PSTN. Just remember phones are FXO and telephone service poviders are FXS and a FXO needs to connect to a FXS and you will be fine :)

What the device sitting between the exchanges needs to do is to receive the call from one and then depending on the number dialed establish a VoIP connection to the appropiate peer. If the device has this sort of dial plan functionality then it will be fine (The Sipura SPA-xxxx devices dont for example). If it does not have this feature you will have to dedicate particular analogue lines from the PBX to connect to a specific remote PBX which will mean you will need more interfaces.

I neither have a separate VLAN or QoS on the switches and I dont have any problems. You only need QoS on the switches if some links could become saturated and packets need to be dropped. Since I only have a single file server a user would have to transfer at the full 100Mbps speed from the server while making a call so the switch has to drop packets (we have the PC's hanging off the back of the switch built into the phones). Practically this would never happen.
If I had multiple fileservers or a server with a gigabit interface then yes I would upgrade the switches so that they support QoS.
For the internet connection I dont use QoS either but instead have a dedicated SDSL line for VoIP. This is better that QoS because with QoS on a router you can only prioritise outgoing traffic and not incoming (thats up to the router the other end of the link).
excelacomAuthor Commented:
So this is interesting, I thought you would connect the gateway to the PBX FXO ports and make the PBX believe the call is going out to the PSTN when in reality it gets converted to a VoIP.
You are suggesting to connect the gateway on the FXS side of the PBX (to analog interfaces I presume) and make the PBX believe the call is going to the extension 3xx but in reality is going to the gateway.

Is that correct?
Sorry got FXO and FXS mixed up.
FXO provides the power etc... and in the type of interface on the PSTN and on the PBX which connects to normal analogue handsets.
FXS are normal phones and the ports on the PBX which connect to the PSTN.

The gateway you found and most of the Sipura devices have FXO ports and so will connect to the FXS ports on the PBX (the ones which normally connect to the PSTN)

Sorry for the confusion.
Ignore my previous post (I wish I could edit it ;) ) I did have it right originally.
See http://www.x100p.com/voip_3.htm for a description of FXO and FXS.

Ideally yes you would use a device with FXS interfaces so that the PBX could connect using its FXO ports and behave as if it is connected to a PSTN.

If you are using a FXS device then to make calls the PBX uses DTMF to make the call. In order to receive calls the device and the PBX need to support DID over the analogue interface so the PBX knows what extension was dialed.
If you are using a FXO device then the situation is just reversed.
As you are probably learning now analogue interfaces are not the easiest to work with :)

Here is another product for you to have a look at.
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