Asterisk and FreePBX

Anyone here running the Asterisk BX with the FreePBX GUI?

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Jeff-1.0 
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Fox's Mercantile
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I don't know what BX is (Business Exchange??) But I run Asterisk/FreePBX at home (which is also my home business).

Wow, the setup was BRUTAL, there's a LOT you have to figure out the hard way. I run a funny setup, two copper CO lines into the PBX, and then several Snom 300 phones via a PoE switch. One of the copper lines is actually a VOIP modem, but that is what the ISP provided, and I'm fine with that, as this machine has no WAN address, only a local address via NAT. More secure that way.

The DAMN thing is so complex, we really haven't figured out how to transfer or park calls reliably. But, the voice mail works great!

So, what did you want to know?

Jon

Reply to
Jon Elson

It's a typo because I make mistakes. ;-) I meant PBX.

The FreePBX forums are almost as brutal.

I want to replace my Panasonic 6x16 hybrid system. Two copper lines from the phone company and a third line from a Magic Jack. 16 analog extensions and a door phone.

I'm using the TDM2400P 24-port analog card. Four FXO inputs and twenty FXS outputs. I managed to get the "internal" extensions and the door phone, a Bogen ADP1, to work.

The door phone was a lot of laughs on the FreePBX forum. "RTFM" and my favorite "Read the FreePBX Wiki." Very useful.

The problem I'm having is I can't dial out or receive incoming calls.

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"I am a river to my people." 
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Fox's Mercantile

I mostly guessed that.

OK, that's about what I did. I had a NEC system from the '80s, and the damn thing just kept chuggung along. But, finally, it croaked, and I was already partly set up with Asterisk and some phone extensions. So, suddenly, I needed to get the Asterisk system up and running.

I'd skip the FXS ports, and just get VOIP stations. I got a bunch of Snom

300 basic phone extensions for $5 each on eBay! Some don't have PoE, so you want to be sure to get the ones with that feature. I got a cheap 48V PoE switch and ran cables where the phones would be, in a lot of places using the old quad wire from the NEC to pull the Cat 5 cable in. I already had the Cat 5 crimping tool from TP Ethernet work.

Umm, yes, the whole purpose of FreePBX seems to be to sell Sangoma phones and hosting. If you are not using Sangoma products, the support is a bit less helpful.

Ouch, that is a fairly serious non-functioning situation. It seems rather FEW people are using the analog stuff. I got the FXO cards working, but am still not real happy with the echo cancellation on my real telco POTS line. The copper line that is 20 feet long to my cable-VOIP modem is great, however, sounds like a recording studio.

I know nothing about FXS ports. But, for sure, you want to get a VOIP extension up, then you can try debugging one side of the analog at a time. You can actually use any computer with a headset to run VOIP software.

The DAHDI setup for the FXO lines was not just plug and play, but it wasn't too bad. Then, there were all the connection rule files for dialing and routing incoming calls. I'm sorry, I went through this about 18 months ago, and have started forgetting some of the details. The system has just been running for some time and is just like an appliance, now. The event logging is really helpful when things are not going right, they give a LOT of info on the steps happening when you try to dial out or a call comes in. If you haven't looked there, that's where you should start.

Oh, and there are a FEW things that cannot be done in the FreePBX GUI system

-- now, of course, I am not remembering what they were.

Jon

Reply to
Jon Elson

That's all well and good, except the house is already wired with CAT3 wiring. Only two of the extension locations have an RJ45 connection available, so VoIP isn't really an option.

< phones and hosting. If you are not using Sangoma products, the

Yes.

I wrote:

Yeah, that "feature" is sort of important.

One of which was setting up the extensions_custom.conf file for the door phone. And of course, figuring out how to get the right set of commands to make it work.

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"I am a river to my people." 
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Reply to
Fox's Mercantile

Yeah, kind-of sort-of.

The whole Asterisk configuration act is brutal. It's probably an elaborate test to find the Chosen One.

I'm not experienced enough to see the problem immediately, but here's how I would debug.

First, can the phones on the inside (16 analog + door) call each other? If so, then at least you did something right.

Second, make sure Asterisk sees all the lines. The outbound lines (two from the phone company and the magic jack) probably need to be configured as "trunk" lines.

It is ideal to have some kind of real-time logging to observe Asterisk in operation. Maybe you can SSH into the PBX? If so, you can get the logs directly from Asterisk by running some command...

The point here is to use Asterisk logs to verify Asterisk interacts with all the lines. If you call yourself (the phone number associated with phone company lines), Asterisk should light up. Notice that if Asterisk isn't registering activity, maybe you misconfigured it, or maybe the phone company doesn't forward calls to the line.

Anyway, once you've configured all the lines, you need to create a proper dialplan. This is where things get interesting, but basically the inbound call from some line will trigger some point in the dialplan and you need to specify what other line will be called and connected to this one.

This is too generic, I know and I'm sorry about that - but I really don't have a FreePBX before me now, and I can't remember the exact bits of the FreePBX interface.

Make sure to check all of the options in the GUI - I know that some Asterisk based PBX-es require you to set a kind of permission to call out in some GUI section that is both non-obvious and far removed from the dialplan.

Reply to
Aleksandar Kuktin

It would appear to be that way.

Yes, I seem to have gotten that part right. Internal to internal is working, the Ring Groups are working and I got the door phone working.

I went thought the exercise again from scratch. Step 1. DHADI configure DID. To assign a Direct Inward Dialing number to each incoming line on an FXO port.

Two methods. At the command line on the box (not the GUI) asterisk -rvvvvvvvvvv > /media/usb/call.text or cp /var/log/asterisk/full /media/usb/full.txt

I found one of the problems. Outbound Routes Silly me, 1 234 567 8901 is an 11 digit number. Helps to have "11 digit" in the dial plan.

Don't apologize, all information is good.

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"I am a river to my people." 
Jeff-1.0 
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Fox's Mercantile

Well, than you are in the finnish line. :)

YES! THAT! It was at the tip of my fingers. ^^

Yes! Outbound Routes. There is also another one. DID and/or DOD. I remeber we needed to set them properly, but that was for an all-VOIP instalation. We use SIP, it carries metadata about the call, but maybe the analog line also carries the same information, just in a different format?

Reply to
Aleksandar Kuktin

An analog telco line does NOT include DID. They rightly assume, "It's a physical line, there's a tag at the demarc telling you what the number is (and always is.)

The way around that is DAHDI Configure DID. You can assign a "fake" DID to an analog FXO port. This way the PBX thinks an incoming call has "xxxx" for the DID and does the Inbound Routes using the DID like it's supposed to.

I suppose if you have more than one incoming line and you pay extra for a hunt group, you would assign all the incoming FXO ports the same "fake" DID using the primary number DID.

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"I am a river to my people." 
Jeff-1.0 
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Fox's Mercantile

Well, you are a bigger expert on this than I. ^^ I'm afraid I dumped most of my internal knowledge, and from this point on I can only look at the interface and try to figure out the solution.

check lines/channels - done set up the dial plan - done fix those DID/DOD things - done

Reply to
Aleksandar Kuktin

One of the problems is using a Magic Jack for one of the phone lines.

From elsewhere: As to your MJ, then depending on your DAHDI hardware, then it might not provide an acceptable ring voltage (90v AC 30HZ expected) less than 30 and it likely wont be recognized as a legitimate FXO

Always something.

--
"I am a river to my people." 
Jeff-1.0 
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Reply to
Fox's Mercantile

I found out a few things to make it work. you HAVE to assign a Direct Inbound Dialed number to make incoming calls work.

Magic Jack does NOT comply with AT&T/Bell ring standards so the FXO port never "hears" it ring.

And because I'm a stubborn bastard, I made a Cisco 7912 VoIP phone work with the system as well. So I can have a phone at the shop.

--
"I am a river to my people." 
Jeff-1.0 
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Reply to
Fox's Mercantile

It gets better, I have 6 of them now AND a VoIP phone line as well.

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"I am a river to my people." 
Jeff-1.0 
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Fox's Mercantile

And because I have way too much free time, I'm configuring five Cisco 7940G two-line phones.

--
"I am a river to my people." 
Jeff-1.0 
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Fox's Mercantile

Well, reading some Asterisk forum messages, I decided to steer away from any Cisco gear, worried about license issues. There are TONS of other phone makes that don't want licenses to use gear you have BOUGHT. I'm using Snom phones, very basic, but do all I need them to do.

Jon

Reply to
Jon Elson

These are the same forums where some self appointed asshole will tell you that Cisco phones are shit because THEY don't know how to set them up.

--
"I am a river to my people." 
Jeff-1.0 
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Reply to
Fox's Mercantile

Well, ANYTHING you read on the internet can be the result of ignorance, or even a hidden agenda! You have to take that for granted!

But, I can usually find useful info on how to get almost anything working after just a few articles.

Jon

Reply to
Jon Elson

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