Connect private Allstar node to Freepbx Box through Iax2 trunk

I have my allstar node 1500 which is in private network with ip 192.168.0
104 and has a hytera radio connected through dmk URI. I want to make a connection with Freepbx server in the same LAN that has few sip extensions i.e 201, 202 etc.
I want to make a call to the radio node from my extensions in freepbx server so that the extension dials the node number and talk to radio through ptt command i.e *99 and #. Also need to view the recordings of these in freepbx server.
I need urgent help as I am stucked at this for so long now.

You need to create a connection between both boxes.
Preferred to be IAX, but you could do it with sip if you cover your security holes.
There are many ways to do this and none of them are really wrong, with exception to your exact needs.
This is just a example of methods


[RPBX] ; Setup for outgoing calls - autopatch trunk to pbx3

So you would need to make a user of the same in the iax radio box. It becomes the peer you are working from.

(keep track of your context.)

Somewhere in your extensions.conf
You make a pattern match and command what to do.

(uses a prefix of 6 to say it is dialing in other box which is removed from the extension to dial “:1”)
So it would dial 5 digit node numbers in other box
Your dialing string must not conflict with anything existing.

The box you are dialing needs to receive this just like any phone connection in specified context.

exten => 29261,1,Answer
exten => 29261,n,Wait(2)
exten => 29261,n,Playback(rpt/node)
exten => 29261,n,Playback(/var/lib/asterisk/sounds/rpt/nodenames/29261)
exten => 29261,n,Playback(/etc/asterisk/msg/idmsg)
;exten => 29261,n,SayAlpha(hf)
;exten => 29261,n,Playback(remote-base)
exten => 29261,n,Rpt,29261|P

I have not used freepbx in a decade, so I am not sure it’s particulars anymore.
As I remember, you can make changes in the conf files but all the real info in in msql DB’s
I prefer to just set it all up manually, so I dumped FreePBX long ago. Kept the whole thing smaller and uncomplicated. You might also when you get accustomed to it all.

That should get you started and perhaps with more questions.
I will say I think the internet is full of help on making iax peering boxes/systems.
Most any methods you see will work, but watch the context you are working from.
Most often you can cross contexts if they are specified in some manner.
When you dial a number, it must be found in the current context by default.

\So that is some general need to knows.

Don’t forget to reload the dialplan after changes.
From asterisk CLI (asterisk -rvvv)

Hopefully my memory is working OK for this.

My method shown above I believe required a registration string in iax.conf from other box.
I would just put it just under asl registrations


Some additional background.

Originally, I had many boxes both PBX and Radio/ASL servers. I call them radioPBX’s
I used a 4 digit extension theme.
Radio boxes used extensions starting with 5xxx
Phone Boxes extensions started with 6xxx
(keeps conflict with 4 digit node numbers down)
I used the last two digits to indicate location for me.
It made it easier to remember for routing and to know what you are looking at much later when you have to debug for some reason.

Today, I use a cheap DID sip trunk and ‘8-line linksys phones’. One or two sip lines per box.
$1 per month but metered for usage.
A couple use iax did. But those cost me $5 per month with unlimited inbound. Great for some things like dialing in to just listen all day when I’m away.
I also use asterisk as an interface to control many things via network controlled interfacing for the repeaters and homes. So, I did have a bit higher need to take it above average.

I know the above is not directly answer your question, but it is food for thought in yours and others reading in future planning.

Make your current thoughts work for now. Ask questions to items you don’t understand.

So, my requirement is to recieve a call on radio box instead of dialing out…what configurations are required for that?