Now what?

Today I threw together a system (Athlon xp64 4200 dual core, I think) and started in with the CentOS 5.1 install installations, modified for CentOS 5.4 with Asterisk 1.6 and FreePBX 2.7.0. I got stuck and ended up switching to the YUM install from the digium.com site for part of it, but everything appears to be working.

I have an OpenVox A1200P with three FXS cards (1-3) and three FXO cards (5-7).

I also signed up for an account with Vitelity.

To start with, I just want to have Asterisk sit in the middle of our three incoming POTS lines and three standard analog instruments, plus be able to make outgoing calls on the VOIP.

I’m looking for basic but thorough instructions on how create the three extensions and three trunks. Something that will tell me when I select “Add ZAP Trunk” what all the boxes mean.

I’ve seen some excellent information, but nothing quite this basic. I guess I’m looking for a document that is so basic I couldn’t stand to use it a month from now, but I’d like to make some phone calls during that month!

Is there such a document out there?

Van

First, I’d recommend a complete solution like PBX in a Flash, Orgasmatron, or Trixbox.

Read this:

http://dumbme.mbit.com.au/piaf/piaf_without_tears.pdf

I put up the website when I was learning hence the name “newbies”. As I found out how to do things, I posted the info. There’s good info on how to create and troubleshoot zaptel (old) and dahdi (new) trunks. PiaF is based on FreePBX so a lot of information is transferable.

PBX in a Flash for Newbies

Another excellent resource is Ringdale’s PBX Administration Guide by Cliffster. Combine that with the PiaF without Tears pdf link that kenn10 provided and you’re covered.

No thanks! Well, I tried Trixbox. I ran into networking problems so I asked a simple question on their forum and was told in no uncertain terms to do it their way for “security” reasons. I decided to pass on taking network security advice from unknown users on a telephony site. With FreePBX I have the public side of the PBX routing through the firewall and the private side directly attached to my local network, without any of the overhead of NAT, port forwarding, and VPN that they were enamored of.

Van

Thanks much for your PiaF for Newbies site, I made a lot of progress based on that.

I’m still having issues stemming from the large amounts of information for Zapata devices and the fact that I’m using dahdi devices. Or at least hope to. dmesg shows that my card is being recognized, along with all six modules. I’ve created three extensions and three trunks in FreePBX to match my three FXS and three FXO cards. Unfortunately, when I do a “dahdi show channels” in the CLI I see my three FXOs neatly “In Service”, but no trace of the FXSs, and I have no dialtone.

Any suggestions on docs to help work through that?

Van

Vanhorn -

Not placing a PBX on the Internet is advice that is generic to any distribution.

It is the same advice that is given in this forum.

If you have a router or firewall that is capable of access lists and can limit traffic then a public IP is acceptable.

A completely cogent response was given to you in the trixbox forum by myself and another highly experienced administrator.

Both of us run Asterisk in production tier 3 data centers so we are not talking out our arse’s.

For reference your post at trixbox.org:

http://trixbox.org/forums/trixbox-forums/help/networking-two-ports

Now that that business is out of the way, why don’t you post all of your DAHDI config files from /etc/asterisk and we can help sort out your configuration issue.