Can't configure POTS nor register phones

Well I thought this would be easy. What I want to do is to replace an analog system on its last legs. Just 3 pots lines in a SOHO environment. Eight or so multi-line phones, corded and cordless, are scattered around the premesis. For now all I want is to replicate what I have. All calls can get routed everywhere. Features will come later. SIP will come later. Again, for now just let’s get these extensions working.

So, I plugged in an AEX844B card and installed freePBX, first manually but finally from the pre-built package. Many times! Currently at FreePBX 16.0.40.4 with Asterisk 19.8.0. Have System Admin and Endpoint Manager too.

I’ve added a couple of extensions. Added a couple of users. Tried to define inbound routes (just ANY). When I try to register a phone using a user/extension it just eventually times out. The log says “No endpoint exists”. Endpoint manager doesn’t see any of the added extensions. No span appears (if it’s supposed to).

The Analog DAHDI Configuration tab does show 4 FXO and FXS ports so I suppose the card is working.

Anyhow, clearly I’m in over my head here. I’m willing to pay someone to help me to configure, though commercial support companies I’ve looked at don’t seem to want to deal with a small SOHO business (and I can’t afford their rates anyhow). I feel I’m reasonably computer proficient but there’s something I’m not understanding here! Any suggestions or any ideas for support?

Thanks all, of course.

two files control dahdi’s setup

/etc/dahdi/modules

says which modules to load

/etc/dahdi/system.conf

for how your channels are defined.

(I don’t use the dahdi helper module, but all my dahdi’s work without it’s help;-) )

good tool to use is dahdi_tool which will dynamically show properly configured fxo’s and fxs’ being opened/closed.

Analogue lines with an analogue card are static; there is no registration. It sounds like you tried to apply a SIP operation to an analogue line. (Although FreePBX uses words differently from Asterisk, I’m not aware of it doing so for registration.)

Hi David55,
Thanks. I’m not trying to register an analogue phone – they’re Yealink IPs phone but I want the analog lines to ring on them. I think (hope) that’s possible :grinning: or I’ll be very disappointed.

OK, thanks. That brings up the question: do these files have to be configured manually or does freePBX work its magic when everything else is configured?

I tried going to Connectivity → DAHDI Config and received a warning stating several files would be erased if I proceeded. In one iteration I tried that, and the partial existing configuration was wiped out and from then on I received a warning on the dashboard about a dahdi config file being unwritable, so I figured that wasn’t the way to go.

Don’t be disappointed, be knowledgeable ;-), Asterisk is a B2BUA (Back to Back User Agent, your Yealinks will be SIP but your PSTN trunks will be FXO (using FXS signalling) You will need to set an (Inbound ) Route from your PSTN FXO port to bridge to your SIP Yealink extensionso it will ring when the FXO is called.

That’s why I don’t use the ‘helper’ module, I don’t find it to be so :wink:

Your Yealink problem is a SIP problem and independent of the POTS side.

Right, that’s what I’m trying to do :wink:. To do this, do I need to manually edit DADHI config files? I do have an inbound route set – basically any calls call go to ring group 99, which rings every extension. But how do I link this route to the FXO ports?

Thank you for your patience. I’d love to find a tutorial but have struck out thus far. And paying someone to help definitely is not out of the question!

Thanks for the thought. I’m trying to digest that one, perhaps it will become clearer if other things get settled.

Log into the web interface of one of the Yealink phones and report whether it was configured correctly, incorrectly or not at all by EPM. For this purpose, looking at the Accounts tab, Account 1 should be sufficient.

What, if anything, appears in the Asterisk log when the phone attempts to register? What does the phone show for registration status? Does the phone display the correct date and time?

Thanks for your thoughts, Stewart.

EPM doesn’t do anything. When I go to Extension Manger on EPM and try map an extension no extensions show, though I have defined a couple of extensions and users.

If I go to the web interface of the phone and try to register the phone manually I receive a "Registration Failed: message.

The logs show
res_pjsip/pjsip_distributor.c: Request ‘REGISTER’ … failed … No matching endpoint found
res_pjsip/pjsip_distributor.c: Request ‘REGISTER’ … failed … Failed to authenticate

The failed to authenticate message is curious. If I enter a user with a bogus password the phone’s web interface immediately returns the “Registration Failed” message. If I put in a correct/user password combo then it takes longer to get the “Registration Failed” message. But either way the “Failed to authenticate” message is in the log.

The phone does have the correct time and I’m able to manually configure buttons etc. through its web interface, though I guess that doesn’t really prove much, of course.

In the Yealink Account page, make sure that both Registrar Name and User Name match the extension number shown on the FreePBX Extension page for the extension you are trying to configure.

Make sure that Password on Yealink matches Secret on the FreePBX Extension page. If your Yealink is really old, it may not be able to handle the long passwords autogenerated by FreePBX. Manually type a password consisting of no more than 12 letters and digits into the Secret field, then copy/paste it into the Password field on Yealink.

Also, confirm that the Extension page shows " This device uses PJSIP technology listening on Port 5060 (UDP) …"

If you still have trouble, post screenshots of both the Yealink acount page and the FreePBX extension page.

when dahdi is correctly configured , dahdi_tool will show it so.

OK, one issue solved, thank you. I was assuming I had to add the extensions as DAHDI extensions since they eventually will talk to the FXO ports, but that makes no sense now that I think about it. So now a couple of phones are registered and can call each other internally.

I still have to figure out how to define inbound and outbound FXO/FXS routes but I’m happy to be one step closer. Thank you very much.

(Though I have a feeling I’ll be back about defining those routes. I need to take care of other things and will continue to search for documentation about that in the meantime.)

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