First time user - USB phone card?

I’m wanting to setup FreePBX to be used at home if possible. I’ve never used this before and it’s kinda alot ot grasp so hoping someone can help guide me a little.

I’m having a phone line wired up next week and wanting to run that into the PBX server to use for calls. So in terms of connecting to the PBX box, am I able to use an rj11 to USB device with the system? If so what are compatible models to do this? I can go with a PCI card but I’ll have to move the PBX box to a different server to do that so hoping to avoid that if possible.

Once there’s a phone uplink to the box is there any additional configuration that needs done on the PBX system to allow calls to be processed through the POTS line?

Long term I’m hoping to setup the PBX box for SIP clients if possible so I can handle calls on my desktop whenever I’m home working and also allow it to be my voicemail service. Also may look into automated fax down the line if everything else works out.

No offense but you may be over your head here… I’d suggest you start with this video series… It’s a version behind but it’s just the basics so it’s still relevant.

Thank you for the video, I wouldn’t say I’m in over my head necessarily. I work with Mitel with SIP/PRI on that system. I’m just asking specifically what are my options for hardware to terminate POTS to the FreePBX system

Browsing over the video he doesn’t really touch base on specific options for POTS hardware, just that it’s legacy and doable with FreePBX.

I’m more so asking hardware specifics for the analog line. Could I run something like the Sangoma - U100 USB with 2 ports? Or am I going to be better off using a PCI card for analog lines like the OpenVox A400E04?

I’m hoping to avoid buying a full blown hardware appliance like the VEGA-60GV2-0800 but if that’s the easiest long term solution it is what it is and I wouldn’t mind that option.

Just hoping for some good feedback on what others have used or known to be usable. Worst case I buy the adapters and see if they work but if I can skip that and get some verified answers if it does or doesn’t work that’d be extremely helpful.

Thanks!

Unfortunately I can’t help you there as I haven’t used that equipment in years. Sorry.

Why don’t you just get a sip trunk? Analog is a dead tech so it’s inevitable that you’ll have to switch sometime.

I suggest you forget the usb stuff, and forget the pci stuff because they are both way past their prime, look to FXS/FXO sip ATA/GATEWAY’s to suit, Sangoma, Grandstream, Yealink, Audiocodes, Cisco and more , that way you are not bound to any particular Server.

Gateways with one FXO (and one FXS that you might find useful) include:
Poly/Obihai OBi212 (current) and OBi110 (old), Grandstream HT813 (current) and HT503 (old), Cisco/Linksys SPA3102 (old) and SPA3000 (really old).

Example:

Yeah the gateway is the way to go to get your feet wet as you don’t drop your POTS line until you feel comfortable and move to a SIP connection. When you get the urge to try a SIP just go with a SIP that uses registration this way you don’t need to get a static IP. I placed my FreePBX behind a firewall without any real issues and without port forwarding.

Here is a link on how to setup FreePBX and a OBi110:
https://support.digium.com/community/s/article/FreePBX-OpenSource-Project-Configuring-an-Obi-110-Device-as-an-FXO-Gateway-i-e-to-allow-FreePBX-to-use-a-regular-phone-line

I have a couple of OBi110 sitting around which I can’t really use any more. I’ll message you and if you want one you can send me an address.

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I can’t believe that a guide written in 2023 would use chan_sip. Not only that, there is no port specified in the OBi (5060 is implied), so it’s assumed to be either a really old system (with chan_sip bound to port 5060), or a newer one modified to put chan_sip on 5060.

Assuming settings on the OBi end are as in the Digium guide, try these for your pjsip trunk:
Trunk Name: (same as AuthUserName in the OBi)
Outbound CallerID: (see Digium guide)
Secret: (same as AuthPassword in the OBi)
Authentication: Both
Registration: Receive
Match Inbound Authentication: Auth Username

Are you sure that what you are getting is really analogue? E.g. I think in most of the UK, if you request a new voice line, you will get a SIP service with an operator provided ATA. The current schedule is that everyone will be forced off analogue in about 18 months, but most places are already “stop sell” for real analogue lines. I imagine most of the Western world is similar.

Yeah I thought that was odd to see a POTS being orderable…but I don’t know where he lives. On the east coast Verizon will not install a POTS line. My dad however in the midwest can only get POTS and for his “high speed” internet its ADSL.

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