My story!

We have a new office where I had 24 outlets for phone that could be used for internet if needed (we did two cables per outlet – one for data and the other for voice for a total of 48 runs). All these cables were terminated on to patchpanels – 24 on one patchpanel and the other 24 on another. I got a Linksys BEFSR41 router and subscribed to Cox cable which provided analog phone lines and internet 2Mb/2Mb connection.

Installed two switches, one each for data (data switch) and voice (voice switch). One port from the router went to one switch and the other went to another. All the connections were tested and all ports are live and internet works fine.

We have two pstn lines – one mainline and another one for faxes but could also be used as voice lines. One of the ports on the voice switch is hooked up to the fax machine. I got Comswitch 7500 which routes the faxes to the fax machine and sends the voice to voice ports (later on this).

I wanted to get a pbx to route phone calls and wanted to check out what’s out there. I am a civil engineer but am always fiddling around with newer technologies. So, I thought I will look into voip. Talked to Fonality and other vendors and for about 6 phones, the quotes I was getting were around $4k plus maintenance charges. Now, I had absolutely no clue about voip. So, I started researching and found many interesting experiences and other expert articles that motivated me to go my own way.

Didn’t want to spend too much money – however, I wanted to get my feet wet. So, I purchased a preinstalled asterisk/trixbox/freepbx from ebay for about $450. It came with 4 FX0 ports (DGM…). Purchased three Aastra and three Grandstream 2000s and hooked them up to the outlets. The two pstn lines (one of them was coming from Comswitch – so the voice line was transferred to pbx) were connected.

I started the box up (I had the pbx box upside down and struggled for a couple of hours as it wouldn’t work – then finally found that out and made sure the pots lines were in the top two analog ports on the pbx) and also the phones. I configured everything in a very basic way- set up extensions, trunks, inbound and outbound routes. The phones just wouldn’t register. I spent all day trying to figure this out – all phones were able to automatically register and obtain Dynamic IPs, but in the Freepbx UI they wouldn’t show. I came back the next day and set all the phones back to default settings, just in case I may have changed some. Then, disconnected them all and connected one at a time and assigned static ips. Bingo! they all registered! I then created a ring group to ring all phones.

Now, the problem is that I couldn’t get phone calls nor could I make phone calls. I tried and tried various things. Eventually, the following setup worked.

Trunks –
Zap/1 (for the main line)
Zap/2 (for the fax line)

Zap Channel DIDs
Channel 1 – Mainline (enter main phone number for DID)
Channel 2 – Faxline (enter fax number for DID)

(Note: When I had the basic single trunk and anyDID/anyCID setup, I was able to use one phone line but I couldn’t use the other line so I set up the Zap Channel DIDs and two separate trunks – I am sure there are other ways of doing this).

Inbound routes (disabled the fax and both have destinations set to ring group – everything else default)-
Mainline
Description (enter Mainline)
DID number (enter main number to match Zap Channel DID on Channel 1)
Faxline
Description (enter Faxline)
DID number (enter fax number to match Zap Channel DID on Channel 2)

Outbound route
Route Name
Main
Dial Pattern:
911
1NXXNXXXXXX
[2-7]XXNXXXXXX
Trunk sequence
Zap/1
Zap/2

At this point, everything was working great.
I wanted to take this to next step of using VOIP provider. So, I signed up with Vitelity and got a local DID from them. This drove me crazy.

I setup everything as suggested on their website (two trunks – one inbound and the other outbound and made all entries as shown for trixbox settings, etc.). I tried various things – changed the sip.conf and also extensions.conf as they indicated on their site for asterisk. I kept getting “all circuits busy”. After hours and hours, I found that I had externip and localnet settings in my sip_nat.conf (that I had added since I though the router had firewall enabled or whatever). I removed them and it worked like a charm. So, if you are using trixbox, the only things you need to worry about are the trunk settings as indicated on their website. Don’t bother editing anything on the config files (I removed my changes).

The route settings are as below:

Outbound route
Route Name
Whatever
Dial Pattern
8|.
Trunk Sequence
Vitel-outbound-trunk

Inbound route
Route Name
Whatever1
DID Number
Number provided by Vitelity
Fax Extension
Disabled
Destination
Ring group

When you click on outbound routes in Freepbx, vitelity is first and my pstn is next (in case you change the dial pattern – right now they are unique so it’s ok).

On the router I had enabled appropriate ports (got this info from some website) and the only change I made on the Freepbx configs was in rtp.conf file as follows:
rtpstart=10001
rtpend=20000

The whole setup is up and running now. I need to tweak many things, but the basic stuff works – took a total of about a week (on and off – ‘on’ most of the time) from scratch to get to this stage.

Cost:
ip/pbx - $450
3 aastra and 3 grandstream 2000 phones - $850
comswitch - $100
switch - $100 (8 port – will do for now)
patchpanel, etc. - $100
phone (two centrex pstn lines)/internet (2Mb/2Mb) service - $165/mo.

Finally, my time – priceless!

Congrats,

all looks good. But there seems to be a line item that is missing in your costs. Given all the savings, where is the $ Donation contribution back to the FreePBX project for making it all possible:-)

(sorry - had to add that)

anyhow congrats and welcome to the community!

Thanks to all in this forum and others. Philippe, the missing line item in the costs has been addressed! Keep up the good work.