Troubleshooting a mystery system

Hello,

I’ve been tasked with troubleshooting an old PBX phone system that I don’t know much about. The system is a CentOS v5.9 with asterisk v 1.8.20.0. The story is that the old IT guy installed the system years ago but has since passed away. It broke recently and we are feverishly trying to fix it, but I don’t know enough about PBX systems to really be helpful. We are talking about replacing this old machine, but in the meantime, I’d like to get it back to a running state. I would also not mind learning about this system so any future replacements or tweaks could be a little less painful.

The system has Elastix, but that seems like a dead project. We were trying to update the OpenVox drivers for the D120P/D120E E1/T1 card and that’s when I discovered asterisk installed on here and started searching around to find this forum. So there’s a community, which is great! Now hopefully there’s a solution -

What should I start with posting to troubleshoot the issue?

Well the very first thing I would do is move to either Alma 10 Or ROCKY Linux or 9 if you wish. then Step 2. would be to upgrade your Asterisk. to 18 or 20 Then all will work itself out. if you have any concerning questions feel free to email me at mejorleads@ Gmail.com

I consult with few providers who offer managed asterisk hosting and setup, I am sure they can help extract the config and setup a managed instance if thats your goal, otherwise you ll need to learn some basics to go about managing and migrating it. Youtube has tons of helpful resources. Start by laying out what you currently have and what working call flow, from there on you can decide to migrate or build from scratch (better way).

I was poking around in the asterisk cli tool. From what I can tell, Elastix is just a web gui to interface with Asterisk, is that correct? Would it be possible, then, to just dump a config or database backup onto a new setup?

Does Asterisk work with Debian? That’s really my preferred distro.

Yes, both 32-bit and 64-bit.

It works great with pretty much anything including Debian. My advice is to get the configuration as is from etc/asterisk and migrate it after installing a new asterisk version.

I believe that there is great chance most parts of the configuration will work with the newer version as is except any hardware device configuration because older asterisk versions used zaptel while newer uses dahdi.

While I was poking around I noticed dahdi a lot. Is it safe to say that this older version uses that already?

It ll be more than just a config dump if you want to do it cleanly and manage in future.

I think I was actually able to fix it by just poking around - I think modprobe and then reloading the drivers and making sure asterisk was running properly fixed it. So with that working (maybe, we’ll dig into more Monday), that gives us more time to make a proper and clean transfer to a new system. Are there docs I can follow for an upgrade like this? I was watching some youtube videos and it looks like there stuff in /etc/asterisk as well as some stuff in /var I’ll want to copy over.

It seems like this will probably strip away an easier to use gui, though. But I’m not sure that’s altogether a bad thing.

Due to the fact that your installation was part of elastix all configuration is pretty much based on it. While it will probably work it would be a pain to maintain it as is.

So I think your approach is best, if you got it working take your time to look into how to use asterisk in the future

Are you thinking to change OS distro in old PBX phone system??
If so, just in case better use a new HDD, to keep old PBX phone system.

OK hold on there. The old IT guy that died did a disservice to the organization with that system by not upgrading it to something modern and not documenting the heck out of it so someone later than him - like you - could come along and pick up the system and keep it maintained.

And now you think that doing the same - replacing a easy to use system with a hard to use one - is a good thing? If I was the business owner who hired you I’d fire you in a second if I found you repeating the same thing that got them into trouble in the first place.

Do this the classy way and drop in a FreePBX system, build it in parallel, then cut the extensions over, and get rid of the internal telephony card and use a FXO gateway like a normal person. Then you can hand over an easy to use system to the person who contracted you in the first place, along with links to paid support from a national firm if you get hit by a bus and die like the old guy did. Do that for the person in charge of the org and they are surely going to keep you involved, keep paying you, and the benefits will be that it’s easy for YOU to maintain.

He died like 5 or 6 years ago. You’re right, though about the upgrade, it would be less of a headache to upgrade this to something easier. I wasn’t sure what my options were, so if FreePBX is possible, then it just makes sense.

On Sat, 2026-03-07 at 02:15 +0000, maxtimbo wrote:

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maxtimbo
March 7

Does Asterisk work with Debian? That’s really my preferred distro.

Yes, but the asterisk version in Debian is derelict.

Debian has way too much ancient software.

It might help to tell what error messages and/or symptoms you’re
getting. ‘Doesn’t work anymore’ isn’t particularly helpful.

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There is a reason for this. It is because Debian 12 is the preferred/mandatory option for FreePBX. When you run the FreePBX install script it adds the Sangoma repositories and installs Asterisk from there not from the bookworm repositories. The Sangoma built Asterisk versions are modern with all security patches.

Nobody except for some very rare border cases is installing Asterisk on Debian from the Debian repositories and the ones that are maintaining some very abnormal systems that for whatever reason can’t be upgraded.

Sangoma sponsors Asterisk development and FreePBX is built around Asterisk.

It is the easiest way to install Asterisk, frankly. And you do not need to use the FreePBX interface if you don’t want to, you can just copy the configs over from the existing system to the custom config files which are not overwritten by the FreePBX GUI interface. Those will not of course be parsed into the FreePBX interface but if something is going on in one of the configs you don’t understand and can’t duplicate in the GUI then you can still use the regular Asterisk config.

A common one would be chan_sip configs, but FreePBX supports those also, just specify Asterisk version 20 as the preferred Asterisk version for FreePBX. There is a migration script to go from those to chan_pjsip if you want.

FreePBX will work in tandem with the old-school way of hand-creating Asterisk configs, it processes it’s own GUI-created configs first then processes the hand-created configs.

Many times people in open source think the GUI interfaces are for newbies but the fact is that if you are building open-source based stuff for an organization sooner or later you are going to have techs far less familiar with open source who will be tasked with maintaining stuff you build. I run an IT department with 4 younger techs and they all eagerly embrace Open Source apps and servers when I wrap them in GUIs but if I tell them “this one has to be configured with nano and a text config file” I get cricket chirps. It’s kind of crazy because most GUI wrappers out there for FOSS servers merely create the textfile configs behind the scenes, but you have to remember the young people grew up with GUIs and it’s easier to row a boat downstream and go with the flow than to fight it.

What I do know is that this is an Elastix gui. I did some lazy googling last night and found there might be a way to restore an Elastix backup in FreePBX. This is a remote job for me. I have smart hands on site, but they’re not very smart. They’ve botched two linux installations and keep getting impatient. I finally gained ssh access which is how I was able to ascertain the architecture. It was pretty wild trying to remember the old way of linux os (mainly init.d). There was also an issue with ssh in that it’s using sha1, so I have to use special args and proxy to secure the chain to me.

You’re absolutely right in that I need to get away from the notion of cli only. But I do need to make sure I have access to this new machine remotely. Currently, they have to air gap this thing unless we’re on a call and we’re working on it. That’s why I’ve been kind of vague about it. Truth is I don’t know enough to say for certain what’s going wrong. All I know is that “it’s not working and we need to install drivers”. Like I indicated before, I think I just needed to reinitialize the drivers for the T1 card using modprobe and restarted Asterisk. But now we absolutely need to migrate this to a modern system with fresh documentation.

I’m “The Linux Guy” in my org. So they call me whenever they see a cli for anything. And I do mean anything - telnet, ssh, bash, powershell.

Arg this is a T1 PRI into the back of a PCI card inside of the PBX thing, isn’t it? For starters how many trunks are coming into this?

I would guess given the age of the system that it’s running an older card that is not supported anymore by Sangoma. Top priority is for you to figure out what card is in there. If it’s something like a Sangoma A116 that’s a 16 trunk PRI card and costs a fortune - it’s worth saving. But if it’s something like a Sangoma A102 - well that’s a 2 trunk card and you can pick them up used for under $100 - it’s literally not worth saving.

I’m very against the concept of terminating a T1 into the back of a PC. 20 years ago I thought that was very cool and I picked up a pile of Sangoma T1 cards that ran a cable into the back end of a DSU and I actually built a router running a T1 into a 80386 Compaq EISA system running FreeBSD and gated, that took a full BGP table from the Internet. I thought it was very cool at the time (I was running an ISP) to replace a $10000 Cisco 7206 VXR with a discarded PC. And it was. But that was then, this is now.

Today the used market is saturated with T1-to-SIP gateways because so many people have gone to data-only to their telephone central office and are running SIP trunks. For example I just checked Ebay and there’s a Dialogics box up there eBay item number:336463739372 that is less than $100. My all time favorite though is the Cisco ISR4321 you can pick those up used along with a T1 PRI card for in the $150 range and easily build a T1-SIP gateway that will run a full PRI with all 23 trunks active. And there’s Cisco techs all over the country, tons of them, that should have no problems with that. Not to mention I just posted a config not too long ago in the Cisco community forums that was a sort of nannyer nannyer nannyer to the Cisco bigots on there showing how to plug in an ISR to an Asterisk system - no UCM needed.

Getting the SIP conversion out of the PBX box avoids nonsense like the following:

“If you are planning to install FreePBX 17 with DAHDI support or are already running FreePBX 17 on Debian 12 and plan to use DAHDI, you must ensure that your system is using a DAHDI-supported kernel version. The latest supported version at this time is 6.1.0-32. If your system has a newer kernel (e.g., 6.1.0-33), you will need to downgrade to ensure compatibility with DAHDI and Wanpipe.”

This is straight out of the Sangoma manual for these T1 PCI cards:

FreePBX Open Source - How to Add DAHDI/Wanpipe support to FreePBX V17 - FreePBX Open Source - Sangoma Documentation

Anyway, if your Asterisk system is speaking SIP to the Dialogics or Cisco ISR SIP gateway, it removes a whole mess of ugliness configuration from the Asterisk system without having to renegotiate the existing PRI over to SIP. In my jurisdiction in fact the state tariffing means SIP trunks are much more expensive than PRI trunks which is why I use gateways.

I would also try to find a local tech who knows Linux. When I was still running my IT consultancy I would have happily taken on a few hour job local support on a Linux box while letting a remote consulting house deal with the heavy lifting.

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