AD6G Allstar Stuff

I've been meaning to get this page going for a while, but lately got some requests for serving a VOIP Phone off of allstar, so might as well get started. In the coming days I'll be posting more of my shell scripts here, so check back from time to time if interested.

I've done my best to remove most stuff that's specific to my setup while leaving some as examples to help you modify them as needed. Hopefully I didn't break anything in that process, but obviously I can't test a script that has strings in it reading "YOUR_ACCESS_CODE_HERE". If you run into problems feel free to contact me.

FYI I'm using HamVOIP so nothing here has been tested on ASL.

Have fun, Larry AD6G


Documentation, notes, ramblings:

SIP telephones hosted by Allstar node


EDIT 2/21/2024 ...Oops. Well, I did say that I might be doing this the hard way, turns out I apparently was. I finally heard from a friend who's been out of town. I'd asked him how he was doing his VOIP phones and if I understood him right, his setup does not route SIP phone through iax.conf like mine's been doing. I'm waiting for him to get back to me with his config. Meantime, what I've been doing works but apparently there are more steps than are actually needed. Derp. I'll leave my old stuff up for now, will probably change when I get the details.

Here's a diagram I made showing how I've set up VOIP phones and ATAs to be served off of my HamVOIP server, giving direct access to the node:
allstar_sip_extension: .odt or .pdf

FYI, I may be doing this the hard way! I'm by no means an Asterisk expert. It seems very cumbersome to toss the call from sip.conf to extensions.conf to iax.conf and back to extensions.conf again, but that’s the only way I could get it to work so far. If you know a simpler way, please do let me know.

Asterisk PBX trunk to Allstar node

Here is a diagram on setting up an IAX trunk from FreePBX to a HamVOIP node, based on a [arm-allstar] mail list posting:
allstar_pbx_trunk: .odt or .pdf

This allows you to configure a trunk access code on the PBX to make connections to Allstar. Node DTMF codes should work; if not or if they behave strangely, you probably need to disable in-call dialing features on the trunk. On FreePBX you do this on the trunk "Asterisk Trunk Dial Options" setting by clicking "Override" and clearing out the dial options field. By default that field will probably show "Tt" - get rid of that.

Needless to say, please be careful that you don't create a hole in your PBX dial plan such that it makes it possible for non-hams to access the trunk.

Shell scripts for HamVOIP

Incoming connection notification

This script monitors incoming connections (excluding some as desired) and can send details to resources such as:

More coming soon

I'll be posting a bunch of scripts I've cobbled together for HamVOIP (as soon as I can get them cleaned up). Nothing spectacular; mostly simple stuff that might save some folks the time to roll their own. Among the scripts are those that:


Free Ham Pages