Do I need multiple Sip trunks if I only have one phone?

I have one sipstation trunk in trial. I need to be able to take 2 calls at once and possibly put a person on hold and switch between calls. I don’t want customers getting a busy or voicemail if I am on one line.

Wiki For Sipstation…not Sip trunks in general.

Short answer - it depends on your ITSP.

I have one trunk through a different provider for the 40 numbers I have coming in.

Call SIPStation and ask, or check their website. That should be pretty definitive

Generally, 2 calls at once mean 2 channels. Generally the trunk is the connection to your SIP provider, On that connection you will have your channels, in your case 1 at the moment. 1 channel = 1 conversation.

Where things may change is for the call waiting. They may or may not require you to have two lines, but very likely if you want two conversations happening simultaneously, then you will need two channels.

PS: If you pay by the minute, then it probably doesn’t matter, you have as many channels as you want. Just have to pay the minutes.

I’m sorry, but I have to disagree.

There are too many variables in the various ways to connect to make a blanket statement like this.

With DAHDI (PRI, T1, individual phone lines), you need one line for every conversation you want to have. This is a limitation of the technology AND one of the reasons I hate it when people talk about “lines” in a VOIP environment.

Within DAHDI (BRI, PRI and T1), you can still have “channels”. These channels are the number of Time Domain Multiplexed slots you are using on your multiple access line. For example, PRI has 23 channels (max) and T1 has 24 (max) and BRI has 2 (max). These channels are handled through the PRI/T1 interface you are using. Regardless, the number of channels you are using on that line will limit the number of simultaneous calls you can send and receive.

All of the channels in a TDM line can be on the same phone number (DID) or they can be individual numbers on “roll over” or configured as individual numbers in a hunt group. This is something you negotiate with your phone company.

You can also terminate individual lines. These are often referred to Plain Old Telephone Service (POTS). Each line is delivered on a single pair of wires. You can have as many of these dropped into your facility as you want - 50-pair bundles (or trunks) are not uncommon. In this model, a single line is almost always a single phone number. If you want two lines, they will each have a designated phone number.

Then, as if that wasn’t confusing enough, we add VOIP trunks. In VOIP, you establish a network connection from a provider to a subscriber. The number of channels used here is not dependent on anything, each channel is used to support a call over the network. The channel is actually a UDP session and is managed as you would any other UDP packet stream.

So, in VOIP, each channel is a phone connection (kind of like a line, but enough not that it doesn’t pay to think of them as the same thing), but there’s no physical analog. Just like with Netflix, you can stuff as many phone conversations down the line as you are willing to put up with.

The place where channels actually becomes an issue is in the “limits” WE place on the connections. As an consumer of VOIP services from my providers, I limit the number of connections I allow at one time. My providers also limit the number of connections they allow to their service (their limit is much higher than mine). We do this to avoid super crazy anomalies (someone breaks into my system and starts 200 calls Dubai, for example).

Since we control the channels we use, we can do it the way we do Netflix (buffering and jitter, making everyone angry) or we can limit the channels based on your outbound network. The limiting factor is codec performance. If you ssume 32kbps for a typical connection and you have a 30Mbps network, you can safely your trunk to over 1000 before you will start causing problems with the calls. If you are on 56K dialup, you should probably set your limit to 1.

Now, like PRI and T1, you can set up a VOIP connection so that it has one DID, or a hundred. This is completely independent of the channels. On a PRI, you can have 23 - on VOIP you can have as many as you’re willing to try to create.


  • In a PRI, if you want to have 23 channels, you have 23 channels all the time even if you are not using them. In VOIP, you only have channels active when you are in a call.
  • In a PRI, you pay for all we channels every month in a set fee. You don’t buy anything additional (except long distance, but let’s hand-wave that for a minute). In VOIP, you pay for your basic connection and “channel-minutes” - a number of cents for every minute there is a channel active in your VOIP trunk.

When setting up Asterisk, you set up a PRI, T1, or VOIP connection (typically SIP) as a trunk. Your incoming and outgoing traffic uses the trunk to get out to your ITSP or phone company. The specifics of how that traffic goes from the line to your PBX is based on processing calls through your configuration of the trunks and your incoming and outgoing routes. In a PRI or a T1 scenario, you don’t get DID information - it’s assumed that you know your own DID based on how you negotiated your connection to the phone company. You route the call based on the line (or channel in a PRI or T1) that the call came in on. In a VOIP call, the number that is being called (the DID) is passed to you with the call, so you need to use that information to make your routing decisions.

On the outbound side, the phone company sets your Caller ID information for you. This, by the way, is why people using Asterisk often wrapped around the axle about which “line” they are using with their PBX - they want the Caller ID to match where they are calling from.

Now, some VOIP providers will set your Caller ID for you, or will enforce a requirement that the outbound caller ID match one of the incoming phone numbers associated with your trunk. Some don’t - some will allow you to send whatever Caller ID you are comfortable with. This is why when people talk about “connecting to a particular line” on a VOIP trunk, we try to unwrap them from the axle - you decide your Caller ID and send it with your call, so there’s no longer a requirement to hit a specific line.

So, in summary.

How many phone numbers are associated with a physical line is up to the people that provide it to you. The phone company will usually give you no more than one number per line (or channel, in a T1 or PRI). VOIP providers, on the other hand, are free to give you as many as they think you can handle. That number limit can be one or a thousand.

For calls, POTS (PRI, T1, or single lines) are one for one. If the line is busy, you can’t get through or you hit Call Waiting. With VOIP, you are limited to the number of channels you have set as the maximum for that trunk. Since you pay by the “channel-minute”, the more calls you place means the more money you pay. This also means that Call Waiting at the provider is no longer pertinent. if you run out of channels (hit the limit for your trunk) you callers will get a message that you are not accepting calls. This, once again, is only dependent on how you configure your PBX.

You might have noticed that I never once talk about “instruments.” In Asterisk, it doesn’t pay to talk about phones when you are talking about trunks. Of course, you need them to talk to people, but your call isn’t going out over a line or a trunk - it is going to Asterisk, which is who is actually talking to the trunk.

You say you disagree, but it sort of reads like your are just restating what I said, just in a much more detailed and exhaustive way. I also agree that calling them “lines” isn’t the best term, but a channel often gets me a blank stare when explaining it. Accountants, Owners, and “the guy who helps with the phones” often don’t get the term channel, so a line is the best term I have.

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looks like I’ll go with sipstation for now. Not sure if these can be ported to another provider perhaps down the road? I still don’t know how many I need to purchase for one phone.

Yes, you can port your number (or numbers) to another provider. People do it all the time.

The problem you are having is this concept of “one phone”. Nothing in the system cares about how many “phones” you have. I have 22 phones in this room, and I have two ITSP’s and 17 phone numbers that provide service to my systems.

The minimum number of anything in VOIP that you need for one phone is “none”. If you only have one phone, you don’t need a PBX. If you don’t want a phone number, you don’t need an incoming number or an ITSP. If all you do is make outgoing calls and know the SIP address of the people you want to talk to, you can connect the phone straight to the Internet and dial away.

Just do whatever you want. It’s clear that you aren’t even paying attention.

sorry to frustrate you as that was not my goal. I understand the phone doesn’t matter. I was more of asking how many incoming calls I can have on one sip trunk. Sipstation said I would need 2 if I want to put someone on hold and answer another call.

In reading about Sipstation, it sounds like they only allow/limit one call per trunk on their SIP service. It also appears as though you can pay additional for their cap busting service which will allow more calls in on a single trunk. I.E. Put one on hold and answer another. It sounds as if they have it setup like good ol’ POTS. I could be wrong, but that is how I read it.

You probably need two trunks for two simultaneous calls with SipStation.

Check out FlowRoute or one of the other sip providers. You can have any number of in/out calls on a trunk up to your network/Internet bandwidth limit. (usually capped a lot lower going up, or out of your network). You can a free trial there too.

With other providers, you may have more than one trunk. The second is for redundancy should something happen to the first.

Good luck!

From experience, UK ITSP’s do not allow free expansion of channels. You pay a per month charge for each channel that you want available, regardless of usage. I’ve not found a provider that charges only for outbound minutes.

I’ve queried this with a couple of them and their response hinged on capacity planning. If I buy 20 channels from them, they ensure they can provide. If I pay them when I make a call, and one day I make 1000 calls at once, they can’t plan the network resource. Sucks, but makes sense.

It’s always a good idea to have a second trunk from another provider to use as a backup in case your main provider goes down or has issues.

Generally, a trunk can handle as many calls (channels) as the provider will allow. Most providers claim to allow an unlimited number of calls, but actually have some kind of undisclosed limit for security purposes. So, you generally don’t need a second trunk just to get a second call

If you have a trunk that allows unlimited minutes, then your provider will likely have a restriction on the number of channels, but you can often increase the number of channels by paying more money without having to set-up a second trunk. I believe that’s true with SipStation.