Two-node (?) Asterisk system, is this practical

I’m putting together an Asterisk system as a pro bono project for a non-profit public-service firm. I’d like to describe what I’m planning and ask you all if it sounds practical.

They need 8 phone lines. For reasons that go way beyond the scope of this email, they can only have three lines (POTS) at their office. So, I’m contemplating a system that looks like this:

At their office: Asterisk PBX with Sangoma A200 card to connect the three POTS lines they have now. About a dozen IP phones in various offices and rooms will connect to this PBX via the LAN.

At the telco (an ISP that also provides telephone and co-location services): Another Asterisk PBX with Sangoma A101 card to connect to a fractional T1 with 8 bearer channels (the 5 they need now and 3 more for future expansion) and one data channel.

The two Asterisk systems are connected together over the public Internet.

The objective is that the two Asterisk systems work together as one, so that, for example, (1) someone could call in on one of the PRI / T1 lines at the ISP and the call would be routed to an office in the firm’s facility via the other PBX, or (2) people could call in to both PBXs via the phone lines connected to each, and participate in a conference call along with some people on the IP phones in the office.

Without getting into the details of how this is done (I’ll RTFM a lot, and figure it out, no doubt with the help of you fine experts here), does this sound like a practical way to do this, and well-supported by Asterisk if configured properly?



Hi yes, But to be honest I would ditch teh pSTN lines and go pure voice, especially at the co-located one. In the office I would as well as voip connections will give them far more flexibility

Good evening, Ian.

We have discussed doing that. There are a couple of downsides, one of which is the co-location facility is 40 miles away in another city, and if the Internet connection between them goes down, all of their telephone service gets cut off. If they keep the POTS lines at their office, they will at least have some communication.

Having the entire PBX at the colo facility raises another issue. The Internet connection at their office is OK but not spectacular… I think it’s 15 Mbps down and 2 Mbps up or something close to that. They want a total of 14 IP phones to be deployed in their facility. If all 14 of those IP phones have to connect to the PBX in the colo facility over their Internet connection, such as during a conference call, I think that will be marginal at best. So I think it will still be necessary to have a node of the PBX at their office to handle all the traffic between the internal phones at the least, and communicate with the node in the colo facility only that traffic that needs to go there.

This may be moot thanks to you for mentioning “FXO gateways” in the other thread… I thought they couldn’t get any more phone lines in their facility but it turns out that may not be the case, and since it appears an FXO gateway might be an affordable alternative to the expensive FXO interface cards for a PC, they may decide to get four more POTS lines from their Internet provider (Comcast) and have all of the PBX functionality in their facility. That’ll be simpler, if nothing else.


For your case I would go with a hybrid system. For a primary phone connection to the world I would use VoIP. If the customer is worried about stability, you can still use a couple of POTS lines for fallback in case VoIP connections is down.

A single VoIP call takes about 100kbps/100kbps bandwidth. So the internet connection speed should be OK. Just if possible, use QoS on the customer router to prioritize VoIP traffic over normal data.

The stability of VoIP connections to the VoIP provider are a very stable thing in my experience …

I have a (sort of) similar situation & I’m wondering if Asterisk is the right tool.

We currently have 8 POTS lines at a doctor’s office. Completely going with voip is not an option, our internet connection is just not reliable enough. We lease space in a hospital facility and recently the internet went out for the whole campus and it took the provider a day and a half to get it fixed.

I’d like to cut down to 2 or 3 POTS lines and do voip for the rest. Problems/issues I see are
-We currently use 4-line phones - is there a good way to keep the system analog inside the office and convert to voip at the server? (Or is there a SIP equivalent available? What I’ve seen here seems to say not)
-The main number is our only published number, with rollover to the other lines. Can this be done with mixed POTS & voip?

I’m sure there are other issues I’m missing, but this is my current state of confusion.

When you say you have “four line phones”, does that mean you have a PBX system at the office or do you just have phones with 4 lines and each one is wired to each line?

As far as the roll over is concerned, that you’d have to take up with your POTS provider. Their switch may not want to do roll-over to a number external to the switch. I see a couple of solutions here:

  1. Get a VOIP DID, have the POTS provider set the last line in your hunt group to roll over to the voip DID. That may or may not work for the reason above.

  2. Get a VOIP DID, keep 1-2 POTS lines, but call forward the main line to the voip DID. This shouldn’t be a problem, you just need to make sure that your telco isn’t going to ding you for long distance calls everytime a call is forwarded. Then, if your internet connection falls over, you can just unforward the main line and work off a limited functionality of 2 lines.

  3. Don’t get a voip DID, just a plain voip outbound provider, use your POTS lines for inbound (leaving the hunt groups alone, just dropping lines), and use the voip provider for outbound. This requires analysis of your call stats, to see if the majority of your calls are inbound or outbound.

Asterisk’s origins are with conventional, centralised, PABXes, not with key and lamp systems. If you have a key and lamp system, you need to think very carefully about what features you really need, and what ones you have just always had.

Whilst I think there is some support for multiple appearance numbers, in Asterisk, it will not be the easiest thing to consider. On the other hand, with a small number of phones, it is quite easy to make Asterisk ring all of them at one, when a call comes in on any line.

Thanks for your suggestions - they are quite similar to the thoughts I was having.

The four line phones are just analog phones, there’s no PBX. The 1st four lines are wired to the receptionists & the nurse’s desk. The 1st, 2nd, 5th & 6th are wired to the business office. The 1st through 3rd and the 7th are wired to the doctor’s phone. The 8th line goes to the fax.

As you can see from the above description, ringing all the lines just won’t work. Currently the receptionists can handle 2, 3 or 4 calls at the same time. Also, the nurse and practice manager can help with the overflow when an influx of calls occur. I can tell at a glance what’s going on with the phone lines.

My feeling is that you are too locked into your existing system to get full advantage from Asterisk. If your analysis is correct, you don’t really need much more than a multi-channel ATA, hung directly off the VoIP ITSP connection.

The factors that might suggest Asterisk are:

  • the ITSP may not be prepared to work that way, and prefer to provide you iwth a single trunk;
  • you can use routing in Asterisk to provide a mix of analogue and VoIP lines.

You would be using Asterisk more like a tandem (intermediate) exchange than a branch exchange.

The strength of Asterisk here is that you can run something like this on commodity hardware, except, possibly for the PSTN line interfaces, and the software is free. The weakness is that you are doing something slightly non-standard, so you will need to learn more about how to configure it than the average small user.

Trying to emulate your existing system using Asterisk and IP phones is not something I would advise doing as an Asterisk beginner. It requires a level of knowledge that will either requiring a lot of poring through HowTo’s, or help from an external consultant.

Okay, another (possibly dumb) question.

I’ve come across the concept of ring groups (I’ve also seen shared call appearance).

Now my confused question - if I have multi-line Sip phones, can I assign each line to a separate extension?

The idea I’m working on at this point is, call comes into DID phone number 1, send it to ring group A which will ring line 1 of the group. If on phone number 2, it goes to ring group B, ringing the same phones but on their second lines.

I’ve been searching but I can’t get a clear picture of whether the various lines on the sip phones can be mapped in this manner. If so, it seems like a workable replacement for our current phone setup as described above (without the SLA drawbacks).

Asterisk allows a many to many mapping between extensions and SIP phone “lines”. The phone may put restrictions on this.

Asterisk doesn’t have ring groups (they an are Asterisk GUI concept). You can define a ringall quueue or you can list the device names, separated by &, in the parameters of the Dial application call.

Thanks for the info.

I’ve tried searching for information on Grandstream GXP2200 (one of the phones I’m thinking about), but I can’t seem to find out what “mapping” can be done to its “lines”

Anyone have any ideas where to find this out (or do I just have to buy a phone and try it?)