Best FXO Gateway for POTs trunks in FreePBX - Gateway Hardware or use Card?

I have a freepbx server running on an old dell in my server rack and a couple POE switch racks here at my house. Id like to have a couple POTs lines coming in to trunk calls out on from within the network.

  1. Is it easier or better to get a card that could handle POTs trunk lines and install it to my server?


  1. What is a good gateway I could use for home/small business use that i has at least 2 FXO ports and maby a couple FXS ports… I dont really need the fxs ports.

Just looking for something easy to get setup from within freepbx. Ideally looking to spend up to $200 and will consider buying used equipment.

It seems as though the cisco spa8800’s are hard to setup or are glitchy? Are the grandstream ones okay? I just dont want any suprises as far as certain things broken or not working well

I have used both cards and gateways and my experience is that cards seem to have less issues in regards to caller ID and supervision disconnection, at least in comparison to the gwateways that I have used.

I have used gateways made by Sipura/Linksys/Cisco, Grandstream, Micronet and Welltech and all have their pros and cons. I would recommend Grandstream GXW4104 if you have more than one line, GXW4108 if you have more than four lines, Linksys SPA3102 or Grandstream HT503 if you have only one line.

There is one use case where you would probably want a gateway over a card and that is when the Asterisk server is running on a VM, as it is not always easy or even possible to make the card work in passthrough mode. If the VM is on the cloud, then definitely you will need a gateway.

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I would go with a gateway, which will continue to function if you move your PBX to a VM, a small form factor PC, a Raspberry Pi, or the cloud. I agree with @arielgrin that GXW4104 is a good choice. HT503 is an older model available used at low cost; current model is HT813. Other 1 FXS + 1 FXO devices include SPA3102 (old), SPA3000 (really old), OBi110 (old) and OBi212 (current).

For more detailed advice, please post: Country? Carrier? How are lines delivered (copper pairs from central office, cable MTA, fiber ONT, etc.)? Internet connection type (FTTH, cable, VDSL, etc.) and speed?

Also, I’m curious why you would want two POTS lines, given that they are expensive in most places and have many limitations compared to SIP trunks. (If budget permits, it makes good sense to keep one POTS line, for most reliable access to emergency calling, for your security system, and as a backup.)

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thanks for the replies guys! good info…

I currently have an unraid server running on a dell r720 i run dockers and vms on and was planning to setup FreePBX on there… imagining a vm would be able to passthrough a card easier than a docker but i may have to look into this.

speaking of which… are there any cards youd recommend if i was to go the VM route? my r720 supports passthough but I’d imagine id have to look further into this in the unraid forums on which cards would work and check to see if this has been done before…

as far as internet service goes, i figured i would upgrade as necessary but currently am on docsys 3.1 cable internet. will probably end up looking into sip trunks but figured i may as well have hardware that could do whatever in the future. I think pots lines from my local provider run about $20.

If you are going the VM route, you definitely want to go with a gateway, it is by far the easiest route.

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So, an fxo is essentially an analog signal , with a bandwidth of 300 to 3000 hz. (most of the energy of a human voice will be in that range) , mostly you want to convert that ultimately to SIP/SDP network traffic.

Consider the rj11 in the wall as the source and Asterisk the destination, there are several routes available.

An ATA accepts the analog and converts it to SIP/SDP in one small box costing between 20 and 100 bucks per channel, asterisk sees it as an endpoint in IP space, you are done.

A DAHDI speaking card takes the analog and converts it into a digium channel that has to get to the asterisk server which converts the digium channel into SIP/SDP , that route must traverse a physical path, the pci bus, via any layers of the kernel(s) necessary to the client server (asterisk) and it must do this with a latency of less than 10 ms and with approaching zero loss/delay or out of orderness anything less than 100% here and you will need to dive into echo correction/detection/suppression which is generally expensive in both time and hardware

Hardware passthrough on VM type systems is very much a variable, does the physical hardware support it or just say they do :wink: ?, do any software layers schedule it against other demands? Many a slip twixt cup and SIP here.

Software passthrough (TimeDomainMultiplexing_over_Ethernet) is a little leaner, you run dahdi on the physical card in the host and then present a layer 2 etherstream to the mac address via a dahdi service running on the target machine, bypassing most parts of ‘virtualization’ that cant wait to bite your ass.

This might sound strange but is the best way I ever could rely on when passing PSTN of analog (or digital, T1, E1, R2, PRI, SS7 or presumably any of the other dahdi technologies) on any hardware running any OS if the PSTN connection and the SIP server where on the same ethernet.

Having worked both ends of this donkey, I can assure you that I would prefer the ATA/Gateway solution for almost any problem, if forced to digium cards, only use the mac layer to get from host to client, it works but prefers its own network interface bare of other protocols, I suggest you use a vlan. ref: (oldie but goldie)



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