Why Do I Need a T1 Card

My new office has a T1 line for Internet. I set up a PBX in a Flash (PIAF) system on an extra Dell computer we had lying around, and hooked it up to the network via my 10/100 ethernet connection on the Dell. I bought a VOIP DID from Vitelity and have done several test calls with great success. The sound quality seems to be acceptable as currently configured.

I would like to start using this system for business using exclusively VOIP DID lines, and connecting internally to VOIP phones via the PIAF box. I don’t plan on using any POTS lines or analog phones.

My question is this: should I buy a T1 card (e.g., Sangoma 101d or Rhino T1)? Will it improve my call quality, or improve the service in any way? Or will I be OK just leaving my PIAF machine connected to the network?

If I do buy a T1 card, would I simply plug my LAN into the card instead of plugging in to my 10/100 ethernet card?

Much thanks for any responses.

Jim S.

The T1 card as you describe is used when you have a dedicated line for telephony referred to as a PRI. In the US, the PRI supports 23 voice channels, in other parts of the world it supports 30 voice channels. You can’t plug a standard T1 line into a telephony T1 Card.

Now if you purchase a PRI, with the proper interface card, you will have better and more reliable inbound and outbound cards.

The PRI is provisioned by your local telco and will have local numbers. You can think of it as a bundle of 23 pots lines terminated into your pbx. But it does cost. We’re paying right at $400.00 a month for ours, or a little less than $18.00 a month per trunk. Standard Business telephone lines run about $65.00 a month each.

If you are using the T1 exclusively to provision the the VoIP box, then you might investigate the possibility of dropping the T1 in favor of a PRI. If you’re sharing the bandwith of your existing T1 between data and and your PBX please be aware that a large file download or high internet usage can seriously compromise your outgoing call quality.

You have to have a relatively large system to justify the cost of a PRI. We have 156 phones up right now, each with DID. Our price has dropped to about $5.00 per phone per month from about $17.00 on the Centrex system since we went to an Asterisk Based System. Our breakeven point for the PRI was roughly 50 phones when we put it in. Since that time, our PRI costs have dropped and now that number is less than 30 phones. Our phone bill has dropped by over $2000.00 per month.

I’m not knocking any of the Internet telephone providers like Vitelity. They do a credible job at a good price, but you are using the internet for call delivery, and things can happen on the internet over which neither you or Vitelity have control.

Hope this all makes sense.


Yes, that all makes perfect sense. We are a very small shop (5 phones) so we would not be a good candidate for a PRI. I am hoping to “split” the T1 into voice and data channels so I don’t have to worry about downloads and other local activity causing any probems. (Still haven’t figured out how to do that yet…)

Thanks for your response.

Jim, what Bill said earlier is totally correct. What happens out on the net is out of your control. However if you have a full T1 out to the net (1.54M) you will probably be okay with some sort of QoS internally depending upon what your switch/router can or cannot do so that there would be no need to spit the T1. That being said I have been running strictly Voip with 15 users with only 2 issues in over a year.

Thanks very much for your response, G711. When you say strictly VOIP, does that mean you are using SIP lines rather than POTS lines both inbound and outbound? If you have time tell a bit more about how you are set up, (what type of internet connection you have, do you use PIAF, what kind of equipment do you run internally, etc.)

If a T1 card is not useful when using pure SIP VOIP, is there any other kind of hardware you would recommend to ensure better service? Or is it best to just hook the PIAF box right into the LAN through the 10/100 LAN connection?

Please excuse me if these questions are very basic. I’m just starting to get my feet wet in VOIP and I find the various standards and protocols a bit confusing.

One of the common misunderstandings is confusing service providers that offer you dial tone over the Internet with service providers that use Voip to transport the call to your premise. The latter is as reliable as conventional telephony, experiences with voice over the Internet vary widely. One person may have a great experience with Time Warner Internet and Vitelity (just using these as examples) in Dallas however the same combination may be unusable in Miami. You simply have to try it for yourself. The problem is the Internet is a best effort service once the data leaves your premise. You can do your best using good routers and firewalls to groom your traffic and prioritize the voice as it leaves your connection however nothing downstream is under your control.

All of the big players, AT&T, Sprint, Level 3 Qwest etc. can deliver virtual trunks over SIP to you on a private T1, these offer perfect quality and great cost savings however they don’t scale down to small installs.

Does your T1 provider offer their own Voip? Make sure it is an in house service and they are not a reseller. Make sure they guarantee, by a Service Level Agreement end to end quality of service and availability.

You should try a few ITSP’s on your T1 and see how it works for you prior to porting your numbers. Using the ITSP for outbound traffic allows you to make an informed decision without putting your business at risk.

Good luck.

Thanks Skyking. I see what you mean about the difference between VOIP to the premises and dial tone over the Internet. My preliminary testing seems to be pretty encouraging. Not a lot of problems with call quality so far using Vitelity. I’m going to try out a couple of other providers as well to see how they do. Regarding my T1 provider (Speakeasy) I know they provide a “hosted” VOIP service but I think that is different from what you are talking about. I will ask them if they have a business VOIP service that I could try out.

Speakeasy will unbundle their voice package. Just ask for SIP termination. If the first person you talk to does not know what you are talking about press the issue until you get someone knowledgeable.

Don’t muddy the waters telling them about FreePBX and Asterisk just ask for SIP termination. If the ask what the endpoint is tell the Asterisk 1.4.

We can help you get the trunk setup.

I am using VoIP for inbound and outband traffic. However I do have a copper line with my published number and I call forward all calls on this copper line to my VoIP number. If my provider goes down then I unforward the copper line so at least I can get 1 call in and 1 call out until the VoIP is back up. As mentioned earlier I have had to do this twice for a few hours. As to hardware, I personally would go with a minimum of a dual core processor and to get further information on hardware take a look at the store on this site and you can get some specifications there or just purchase one. My broadband connection is 10 meg up and 10 meg down with a delay of about 20ms to my VoIP provider, so I’m happy.

To get your feet wet I would recommend doing a mix of VoIP and copper lines. Use the copper for your inbound traffic and use VoIP for the outbound and then if VoIP is working for you consider dropping some or all of the copper or whatever your comfort level is. If VoIP is not working for you then route all the calls via copper. (Please keep in mind 911 calls).

The PBX would be connected to your LAN. I will assume your T1 is connected to a router and the router is then connected to a switch and all computers are connected to your switch. Your PBX would be connected to your switch and hopefully in your switch or router you can put in some sort of QoS either by physical port, ports such as 5060 or by IP address.

I also would suggest doing some home work such as what is the needed bandwidth for 4 calls at the same time using g711 as the codec, what causes echo, what causes jitter, what causes 1 way or no audio, etc… Jump on other message boards and go through what others have gone through such as this site, nerd vittles, tricky-box, voip-info etc. Download or buy Asterisk The Future of Telephony, download Conversational Linux by Joe Roper and even better yet go to the training promoted by FreePBX (well worth the money).

The above are just my opinions, and as you know opinions are like cowboy hats … everybody has one.

I asked Speakeasy about unbundling the voice package. They said they do not do SIP termination, and that they don’t support it. However, (somewhat contradictorily) they also gave me the SIP “password” and username associated with my account. They said that some customers had used this information (on their own) to control their SIP termination.

What now?

Well all you have to do is find out the server IP and you are in business.

You said they are delivering as a PRI to you so you will also have to get the inbound routed to your server.

What did they use onsite for an IAD? If it is Cisco it would be very easy to reverse engineer.


I just have a basic T1 line, not a PRI. Speakeasy is the provider. The T1 is delivered through a Netopia router. I do know my own IP address, but I’m not sure if that’s what you are talking about.

I also ordered a $20 per month home VOIP package which is normally delivered via an ATA (Linksys). I’m not interested in using the Linksys ATA, and was hoping that it might be possible to program my PIAF box to accept the code from the Speakeasy Home VOIP service. Does that seem like it might be possible?