(Sconkel) 1

I am upgrading our companies phones to VoIP (thanks to the advice of some very helpful spicers)! I am gonna run Elastix on a box and have some softphones and then some existing analogs (using some of these: Amazon.com ) connecting to the server. We have 9 incoming POTS lines that we are using to call out.

My question is what is the best way to connect those lines to the server. A single standalone FXO gateway, a PCI card on the server, or something else. I think I like the idea of having an FXO gateway that connects to the network (POTS lines are not near our server room so a PCI card would be difficult).

Just trying to get some clarification before I order some hardware and go to stage 2 of testing (already set up Elastix and tested it on some softphones).

Any suggestions on a good (cheaper end, phone communication is not a lifeline for us) FXO gateway or what a better option would be? I was looking at this one: Amazon.com

Thanks again!!

4 Spice ups
2

That’s how I would do it, but I have very little real-world experience with those.

3

Gateway would be the easiest in my opinion. I think you are on the right track.

4

Use the gateway that way you can virtualize your server for production.

(Sconkel) 5

I currently have 9 lines. If I get an 8 port FXO gateway and I need more can I purchase another 8 port Gateway and stack them or would I have to get a 16 port gateway and scrap the 8 porter??

Also when people in the office call out will they all show the same number on the receiving phones caller ID (we have 9 different numbers right now) or will some calls shows one number in the caller ID and another call show a different caller ID?? I would assume it can be set to show the same ID but i wanted to verify!

ex. John Smith calls ATT and it shows up as 937-123-4567 to ATT and Bob Jones calls Verizon and it shows up as 937-123-4567 to Verizon (or will it show like 937-456-7890)?

Thanks for your help guys, it’s been a life saver!

6

If you are stuck with the POTS lines, then the gateway is the way I’d go. And as Jared points out… virtualize Elastix.

If you can get SIP, you can eliminate the cost of the POTS and of the Gateway and get cheaper calling per minute too, though.

1 Spice up
7

Gateway will definitely be what I go with then. Are FXO gateways stackable if the needs ever comes up?

1 Spice up
8

Sure. FXO Gateways convert FXO to SIP. You can have as many SIP trunks as you like. So you can have many gateways if you want.

9

Thanks! My last question before I quite bugging you all then would be:

When people in the office call out will they all show the same number on the receiving phones caller ID (we have 9 different numbers right now) or will some calls shows one number in the caller ID and another call show a different caller ID?? I would assume it can be set to show the same ID but i wanted to verify!

ex. John Smith calls ATT and it shows up as 937-123-4567 to ATT and Bob Jones calls Verizon and it shows up as 937-123-4567 to Verizon (or will it show like 937-456-7890)?

10

That’s not a simple question. That comes down to how you decide to set up your calling. And the complexity of it is one of the reasons, besides reliability and cost savings, to go to SIP trunking so that you can do anything you want rather than being limited by the POTS lines.

1 Spice up
11

Did you ask your pot lines provider is it possible to make them trunks directly by the provider. I’ve done this by signing a contract and they rearranged their network connections.

12

Basically, no POTS will always show the POTS # to the recipient’s caller ID. Anything else has more options, even a PRI. When you go SIP then you can specify things yourself in general. THere are rules about it, but as long as you own the numbers, you can choose to show the one you want in general.

2 Spice ups
13

I thought there were still timing issues running asterisk as a vm, or has that been resolved? I have heard people successfully running on Xen but Esxi was not as successful

14

That never existed, definitely a myth from bad configuration :slight_smile: That was only when using horrible Type 2 VMs like VMware Server.

Xen always worked (people without it working didn’t know Xen and screwed up using full virtualization instead of PV) and ESXi has worked for a very long time. It’s absolutely rock solid on all enterprise virtualization platforms.

Xen has been rock solid since the beginning. Their PV is in lock step with the CPU.

1 Spice up
15

Port those numbers to a SIP provider if you can to take the POTS lines out of the loop (unless you just have to keep them). Then you’ll be charged a small fee per DID per month (sometimes as little as $1 depending on your contract terms) to keep the numbers and then a certain fee based on the number of concurrent calls you need. So if you only need to have 5 concurrent calls at any time, that’s all you’ll need to order from your SIP provider (normally referred to as a trunk).

When you move your numbers to a SIP provider, they will offer services for inbound caller id name (you see the caller id name and number for inbound calls rather than just the number) and likely charge you per DID you own per month to have this service. And with outbound caller id, you can make outbound calls from any phone at your location show the same caller id name. There is often times an activation fee for this service as well. But again, it depends on contract terms.