Welcome to crosstalk solutions: my name is Chris, and this is free. Pbx 101 for version 14, video 10, where we’re going to be talking about PBX trunking and we’re taking a step away from the computer to just work on the white board for this video, because the topic of PBX trunking is so so huge right. There’S so much to learn and what I’m going to cover are the most common types of PBX trunking that you’re going to encounter, and that is pots lines, PRI lines and SIP trunking. Okay, so we’re gon na cover all three of those in detail. But let’s go ahead and start with pots lines, so pots stands for a plain old telephone service, and these are the old-school copper lines that you would have, for instance, at your home right back.
When people didn’t have cell phones, what did they have? They had a home phone and that was delivered over a copper pots line. Now, pots lines are still in use today, especially in some of the more rural areas, sometimes they’re the only option, if you have a small business – and you can’t get you know internet, that’s good enough, for you know voice over IP. It might be your only option now, how do you connect a pots line or multiple pots lines to a PBX server? Now that is done with either an analog gateway such as this sangoma Vega, 60g, that I have right here, so this one can do up to four pots lines.
However, they make models like this that can do up to eight pots lines or you can do it with a card in the server, so that would typically be a sangoma card, a Digium card, something that has FX o ports, so analog gateway or a card. This is typically going to be a PCI card or a PCIe card these days with FX. Oh ports not to be confused with FX s, ports which are station ports used for things like fax machines, okay, so analog gateway or fxo card, and the FX. So port cards come in flavors of you know two four 680. I think up to 24 FX, Oh ports per card now the upside to utilizing pots lines is that they’re very, very reliable they’ve.
They don’t go down right, so the old legacy. Copper infrastructure is been around for so long, it’s incredibly stable and it’s just a really solid way to run your phone traffic. However, on the downside, they’re pretty expensive, so a single pots line is gon na cost you somewhere between 20 to 30 dollars per line, and that’s not counting any of the extra features like caller ID or 3-way calling and that’s also not counting any of the actual Usage on the line, so they’re they’re, relatively pricey, and they aren’t very flexible so, for instance, take caller ID. So you can take multiple pots lines and you can string them together in what’s called a hunt group. So a hunt group also known as a rollover group or a hunting group.
Basically, what that does is you’ve got line 1 line 2 line 3, and so, if your main number is line 1, that’s your main office phone number. Someone calls that number. That line is in use if the second person calls in they call that same line. Well, then it’s going to roll over to line 2 and then it’s going to roll over to line 3 and so on and so forth. Until you run out of lines at the end of your lines, if you don’t have any role of Reliance left, the caller gets a busy signal.
Now, that’s fine for inbound calling and it actually works really well. This is something by the way that you have to set up at the carrier. This is not something you can do in your own PBX, if you’re, using AT & T, for instance, AT & T, has to set up your hunt group for you. So, on the inbound side that works really well on the outbound side, however, it’s a little bit more problematic, because if your main caller ID is five five, five one, two one two, that’s the caller ID that you want to be giving out and all your you Know: marketing material, that’s the caller ID that everyone knows. That’S the caller ID that’s in the phone book, but when you call out the first call might go out this copper pots line with that caller ID.
But then, when that line is in either inbound or outbound, the next call is gon na go out line two and it’s gon na go out with line twos, caller ID phone number and then line three is gon na go out with line threes, caller ID phone Number so, as you can see, it’s pretty rare that when you’re dialing outbound you’re gon na actually be out pulsing your company’s desired caller ID phone number or your company’s main caller ID phone number. You can only have that going outbound once so. That can be a problem for some companies, especially as you’re switching over to voice over IP you’re going to have to port all of these phone numbers over to the new service. Because you know, if you call that online to once and someone saved that number as your business’s caller ID they’re always going to be calling back in on that same number, so it can kind of cause a problem when you’re shifting to a newer technology. Now, of course, I mentioned when you run out of lines in your hunt group typically you’re, just out of luck.
Your users are gon na get or your callers are gon na get a busy signal, or you can have like AT & T, set up a voicemail box as the final rollover, but that’s a voicemail box. That’S not in your phone system. It’S not in your free PBX, it’s with the phone carrier, and so then you have to remember to check the phone carriers voicemail box periodically. So it just adds a little bit of extra work. For you know whomever is responsible for checking the voicemail okay.
So next, let’s talk about PR, i –‘s now PR is are also known as T ones, they’re, also known as e1 s. These three things are kind of interchangeable, though they are different technologies. A PRI which stands for primary rate interface is a bundle of lines. Typically, it’s served over copper, but we’ll talk about that in a second, but it’s a bundle of lines where you get 23 voice channels. So you can have up to 23 concurrent calls and you get one D channel or data channel, and that data channel is for passing things like caller ID now.
The older technology is 81. 81 is simply 24 voice channels, groups, together with no data channels. So a t1 isn’t used much these days because it doesn’t have any caller ID there’s no data channel to pass that information along. He ones are also similar, but they are used in other countries. So pris are primarily used in the United States and I believe Japan uses pris as well, but most everywhere else in the country.
Excuse me most everywhere else. In the world, you’re gon na find II won s instead and an e one consists of thirty voice channels and two D channels: okay, so to connect a PRI into your free PBX, you’re, also going to need a digital gateway. So here this is a Vega 200 G. This has the capacity to terminate to PR i –‘s into a PBX, so the pris terminate into here, and then this then connects to your local area network and converts that PRI signal into voice over IP and then puts it right into the PBX. There are also cards for PR i –‘s, just like the pots lines.
They have PRI cards that can go into the server they’re, usually flavors, of one two or four port PRI cards, but there are certainly PRI cards that can get you a lot more than that. But if you’re dealing with that many PR i –‘s chances, are you don’t need me you’re gon na be on the enterprise level? Okay, so the upside to PR i –‘s is that they are also very reliable PR. I –‘s most often use the same copper infrastructure that pots lines run off of. However, more and more often these days, you’re gon na find that PR, i –‘s run across voice over IP until they get to your building and then the carrier converts it to a PRI handoff.
So actually it’s transparent to you. It still seems like a PRI, but in reality it’s R, it’s running sip when it actually gets out into the you know the wild wild west. If you will now, the downside to a PRI is that they can be relatively expensive. A PRI, depending on your area, is usually going to run you somewhere between 250 and 400 dollars per month. That does not count usage.
Okay, so you can see that it’s a bundle of lines, it’s cheaper to have a PRI than to have 23 pots lines, but it is still going to be relatively expensive, come to some of the sip alternatives. Okay, so let’s finally talk about sip and sip stands for session initiation. Protocol sip is a very common voice over IP protocol. It’S not the only voice over IP protocol. So basically all sip is voice over IP, but not all voice over IP is sip.
There are other examples, such as the skinny protocol for Cisco phones or Cisco equipment, there’s also the ia X or inter asterisk exchange protocol, which is a voice over IP protocol. That’S used for connecting multiple asterisk systems together, but by-and-large sip is the most popular voice over IP protocol, and so when someone says SIP trunks or VoIP trunks, they’re used fairly interchangeably. So especially if I accidentally say boy trunks, SIP trunks and VoIP trunks are sip. So with SIP trunking, there is no card needed in your server whatsoever. Sip is very good for hosted PBX systems because again you don’t need a gateway to terminate locally and then send across the Internet to your hosted PBX.
The SIP trunk terminates directly into your hosted. Pbx everything is IP based it’s right on the IP network within sip. There are all different types of SIP providers, so you’re gon na have pay-as-you-go providers that those are providers such as flow route, vitality, voice. Ms, these are providers where you input or you deposit, a certain amount of money upfront. So, for instance, you plunk down.
You know 50 bucks into your flow route account and then, as you make calls it’s just deducted from that $ 50. Now the advantage to a pay-as-you-go SIP provider, especially for smaller businesses, is you can throw a hundred bucks into your account and kind of forget about it until you run out of that hundred dollars credit, and sometimes that’s going to be months right. So you don’t have any sort of monthly recurring charges. You just have an amount in an account that is being deducted from at you know, a penny per minute for every call that you make. Mrc providers are more like your standard atnt like what you would think of as your home phone right, they’re gon na charge, you a monthly fee that monthly fee typically includes some number of minutes.
So, for instance, you pay $ 35 a month for your SIP trunk, but it includes five thousand minutes of outbound dialing. So an example of an MRC provider is going to be sangomas sip station. They charge, I believe, $ 25 per month per channel, and then they also charge you for usage, outbound usage and stuff. Finally, you have your enterprise providers right this, so this is going to be your charter, your Comcast, your level, three AT & T. These types of providers can provide you with SIP trunking.
It’S typically going to be about as expensive. It’S going to be on par with a PRI, but they can usually break it up into like ten channels of SIP trunking or you know, 30 channels of SIP, trunking they’re not limited to just 23 channels, and then you got to get another PRI. They can usually they’ve got some. You know variants in there with enterprise providers. You are paying more, but you’re, also typically getting some level of SLA, meaning that, since they have control of the SIP trunk from start to finish, we’ll talk about that in a second.
They can they can basically guarantee quality and uptime on that line, for you as part of the contract, so for my larger customers, enterprises sometimes the way to go because they need to ensure they need to have that sort of extra insurance that the lines not going To go down, whereas with some of the smaller pay-as-you-go providers, there’s no guarantee that the line is going to be up or down, and is it your internet that goes down or is it the the provider? That’S gone down right. Both of these are dependent on your internet, whereas enterprise typically is a dedicated circuit that is not dependent on any other data internet connection that you may have. So speaking of that enterprise gives you a dedicated circuit, not always, but very often, and that means that they’re going to provide you equipment on site that you’re free PBX is going to do sip peering with and we’ll talk about that more in just a little. In just a second here, so let’s talk about the types of SIP terminations, so the type of sip providers and that’s the type of sort of payment structures that you can expect to see with different sip providers.
However, how do these sip providers actually register to your PBX? In fact, I’m gon na get rid of this too. Okay, so here’s the free, PBX server, here’s your firewall and then this is the Internet where your sip provider lives. Okay, so there’s three different ways that they’re going to that you’re going to register a SIP trunk in order to send and receive calls to the outside world. So the first way is password authentication.
So with a password authentication, your SIP provider gives you some credentials. A username and a password and then an IP address or fully qualified domain name to connect to in the outside world. And so basically, you input that information into here user pass and we’ll talk about the actual trunking and how to configure that in free PBX. In the next video, but you’re basically going to be entering some information that includes a username and password and then free PBX registers outbound through the firewall to the SIP provider, and so it creates this connection here. That is now a stateful connection.
It’S held open through the firewall, because firewalls allow outbound stateful connections and so since you’ve maintained that connection phone calls in goes straight to the PBX and phone calls out goes straight to the SIP provider, so it works. Okay. The problem that I usually have with this type of sip, based or password-based authentication, is that the firewall can tend to be problematic. Sometimes it’s going to sever this connection and then it takes some time for the free PBX to realize that it needs to re-register that trunk with the SIP provider. So it’s not always the most solid way of doing sip registration.
You know it works most of the time, but again it depends on the firewall that you have and it depends on how that firewall is configured. I personally like to eliminate the firewall as a source of the problem, and you can do that with the next type, which is IP based, authentication, okay, so with IP based authentication, there is no need to have any sort of user name and password. So the configuration in the free PBX is typically a lot easier to do. You’Re, basically, just saying hey, send all phone calls to this IP address, or this fqdn. So outbound is just saying: send to send to XX, XX right where xx xx is the IP or fqdn of your sip provider.
Ok, so, basically, if you’re on vitality, you’re gon na send to send all calls to outbound dot vitality net or something like that, and then it just pushes those calls out on the inbound side. You would then configure vitality to send all inbound calls to the when IP of your firewall. So if your, when IP is y dot y dot y dot Y you’ve configured the SIP provider in their online portal to say, hey, listen, send all calls to y dot y dot y dot y, which is typically on the front of your firewall. And then that’s gon na route inbound, you know any calls that are any traffic that comes on port 5060 from y dot y dot y dot y, send to the LAN IP address, ZZ ZZ of the free PBX right. So it basically does that nat translation in to the free PBX, it’s a very solid and stable way of doing authentication.
The downside to doing it. This way is that you have to open up ports – 50, 60, UDP and 10,000 through 20,000 UDP in your firewall. You do have to open those ports, however, you can lock those ports down to the SIP provider. Well, I guess this is the supervisor you can lock those ports down so that you’re only allowing traffic in on these ports this IP address. Okay, so that’s the way that I prefer to do it that way, someone would have to be a network genius in order to hack your sip account they’d, have to know your IP they’d have to spoof your IP.
They have to know the IP of your sip provider. They’D have to spoof the IP of your sip provider and then they’d have to you know, know about all your inner workings and you know how everything’s routed and all that sort of stuff so very difficult to bypass the security here if you’ve locked it down properly. But do not ever open these ports through the firewall to the entire world? Okay, so fair warning. Never do that because you will get hacked almost immediately, not necessarily hacked, but they will have.
People are running scripts constantly on every IP address in the internet. Trying to find vulnerable voice over IP servers that they can make free phone calls on okay. So if you’ve locked yourself down to only the IP address of your SIP provider on the inbound side, then you’re going to be fine, typically, okay. So finally, I want to talk about sip handoff. This is also sometimes known as sip peering, but basically, when you have an enterprise-level provider that is giving you a dedicated circuit specifically for voice over IP, this is typically how it’s going to work.
It’S not going to use the general internet data connection, that’s connected to your firewall, so you’re free PBX, for instance, when it goes out to get updates – and you know, package updates module, updates, it’s going to be downloading through the firewall and getting that stuff from the Internet, but when the free PBX makes phone calls sends and receives phone calls, it’s actually peering with a different box, so AT & T or Comcast or charter or whomever the enterprise provider is, is typically going to put a piece of hardware on site and you know Usually it’s like an ad Tran box or something like that, and that is a dedicated circuit with their home office. Okay, so often it does run through the internet as well, but it’s a dedicated circuit, separate from your data internet circuit and they’re, going to give you a different IP address to use so usually to be like 172 dots, 16.1, 23.2 and they’ll tell you that you Need to peer with this IP address, or rather with their IP address, which would be like 172, 16, 120, 3.1, so they’re, essentially creating a small local area network or an additional local area network inside between their box and your free PBX.
Now this is typically going to require multiple network cards on your free PBX one for the standard internet connection, which is the gateway. You know you just pass traffic out to the Gateway. You know download your updates and that sort of stuff and then one that doesn’t have a gateway IP, it’s just an extra network that is sort of this sip peering network over here now again, that’s typically only going to be with larger companies that are giving you A dedicated piece of hardware to put in your land or in your network, I should say: okay, so that about does it for SIP trunking, again we’re not going to get too in-depth going to show you every different way that you can configure a SIP trunk in Free PBX in our next video, because there’s just too many ways to do it right. So what I’m going to be doing is a vitality trunk with username and password based authentication, and hopefully that will work just fine for our purposes. Okay.
Well, I hope you guys enjoyed this video and stay tuned for the next video, where we’re actually going to connect up a SIP trunk and start sending out some phone calls to the rest of the world. Okay, if you enjoyed this video, give me a thumbs up if you’d like to see more videos like this, please click subscribe. My name is Chris with crosstalk solutions and thank you so much for watching [, Music, ]