Asterisk and FreePBX

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Anyone here running the Asterisk BX with the FreePBX GUI?

--  
"I am a river to my people."
Jeff-1.0
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Re: Asterisk and FreePBX
Fox's Mercantile wrote:

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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

Re: Asterisk and FreePBX
On 2/14/18 2:44 PM, Jon Elson wrote:
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It's a typo because I make mistakes. ;-) I meant PBX.

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The FreePBX forums are almost as brutal.

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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."
<https://wiki.freepbx.org/display/FOP/Door+Phones
Very useful.

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



--  
"I am a river to my people."
Jeff-1.0
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Re: Asterisk and FreePBX
Fox's Mercantile wrote:

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I mostly guessed that.
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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.
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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.
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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.
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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

Re: Asterisk and FreePBX
On 2/15/18 5:07 PM, Jon Elson wrote:
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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.

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

I wrote:
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Yeah, that "feature" is sort of important.

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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.


--  
"I am a river to my people."
Jeff-1.0
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Re: Asterisk and FreePBX
On Wed, 14 Feb 2018 17:16:34 -0600, Fox's Mercantile wrote:

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Yeah, kind-of sort-of.

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

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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.

Re: Asterisk and FreePBX
On 2/16/18 6:23 AM, Aleksandar Kuktin wrote:
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It would appear to be that way.

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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.

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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.

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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

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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.

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Don't apologize, all information is good.


--  
"I am a river to my people."
Jeff-1.0
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Re: Asterisk and FreePBX
On Fri, 16 Feb 2018 07:05:05 -0600, Fox's Mercantile wrote:

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

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

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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?

Re: Asterisk and FreePBX
On 2/16/18 7:16 AM, Aleksandar Kuktin wrote:
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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.




--  
"I am a river to my people."
Jeff-1.0
We've slightly trimmed the long signature. Click to see the full one.
Re: Asterisk and FreePBX
I just discovered I was using my old e-mail address when posting. This is  
the correct one. The old still works, but is deprecated.

On Fri, 16 Feb 2018 07:38:31 -0600, Fox's Mercantile wrote:

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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

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