Upgrading to version 17 and using billing engine

FreePBX Integration
1

Hey everyone! So I’ve got a bit of a head-scratcher here. Is anyone else stuck dealing with this mess? I’m still rocking version 16 and kinda iffy about jumping to 17. Every time I try to install 17 and do a backup reinstall, I’m just flooded with error pop-ups. Anyone else feel my pain? Is there like a secret sauce for getting a smooth reinstall from backup on 17? For now, I’ve decided to stick with 16 until it’s completely dead. Poor old version 16, though—I’m struggling to find any docs at the moment ‘cause of some annoying issues I’m having trying to restore from 16 to 17. But that’s just one side of it!

So here’s the real deal: I’m on the hunt for a billing engine. Here’s my game plan: I want my users to connect via an extension number and be able to call out using a trunk that’s tied to a billing engine—like Magnus Billing or ASTPP. ASTPP seems a bit friendlier, but I keep running into authorization headaches when trying to get the trunks to chat with each other. Magnus Billing is a bit of a pain too since it’s all about that SIP life, no pjsip for them! I also tried IAX but hit the authorization wall again.

What I wanna do is super simple: when someone makes a call, it should ping one of those billing engines. The billing engine sees the extension number, verifies it (cuz it’s set up on its end too), and checks if we’ve got enough credit to go through. Once it gives the thumbs up, it sends the call back to the PBX, using the dial plan already set in the outbound trunk on FreePBX 16, and bam—we mix the call like usual!

I’m just aiming for that sweet automation, kinda like how ( A2Billing RIP ) played nice with Asterisk on FreePBX. If anyone’s got tips or can point me in the right direction, I’d be super grateful!

2

Do those billing systems support chan_pjsip?

3

Hi, Blaze I think ASTPP dose.

4

It would help to know what issues you’re actually running into.

5

I’m having a tough time setting up an internal trunk between my two PBX systems—one’s Free PBX v16 and the other is ASTPP Billing, which uses FreeSWITCH. They both work great on their own, but I can’t seem to connect them. Every time I try to set up the trunk, I get this “unauthorized” message. I’ve checked everything, and all the login info is spot on. They’re both on the same LAN and subnet, but when I try to set up an IAX2 connection, it just doesn’t communicate with Free PBX. I’m not getting any errors or notifications from Free PBX either when I try the IAX2 setup, and neither system can use the trunk to connect back to Free PBX. So, how do I go about setting up this trunk between Free PBX 16 and ASTPP billing? I’m really stuck and have no idea why I’m getting that unauthorized message. By the way, when I try the same settings with Magnus, I’m seeing the same issue, and Magnus mostly uses standard SIP.

6

Which 'unauthorised message. This is the first time you have referred to one.

Also 401 Unaurhorised is a normal part of the SIP handshake when authorisation is required. If it isn’t required, it generally means that the UAC has not been recognized, and the UAC is trying hid that fact from a potential attacker.

What does this mean? Asterisk (both chan_sip, and chan_pjsip) mostly uses SIP according to the standards for SIP.