Connecting an AI phone agent to FreePBX and Asterisk
Asterisk is open-source telephony software, and FreePBX is the best-known management layer on top of it. You connect Tibinet's AI agent with an ordinary PJSIP trunk. One outbound route and one inbound route make the agent a destination in your dial plan.
FreePBX, Asterisk and the AI agent
FreePBX is maintained by Sangoma. Other Asterisk-based systems, such as Issabel, work the same way. Your handsets, IVRs and queues stay as they are. The agent is added as one more destination.
How the integration works
- PJSIP trunk to the SIP address we give you. Either with registration ("Registration: Send" and outbound authentication) or with IP authentication (an identify on our address).
- Outbound route for the agent numbers 85XX and 88XXXXX. Every agent has a fixed seven-digit number starting with 88 and, on request, a short number such as 8500.
- Inbound route for the transfer numbers you agree with us.
- Channel driver
- chan_pjsip
- Port
- 5060 (UDP and TCP), 5061 (TLS)
- Media (RTP)
- UDP 10000 to 20000
- Codecs
- alaw (preferred) and ulaw; G.722 on request
- DTMF
- rfc4733
Every setting on our side is listed on the technical details page.
What your administrator does
Create the trunk
In FreePBX, go to Connectivity, Trunks and add a SIP (chan_pjsip) Trunk with our details.
Create an outbound route
Add an outbound route with the patterns 85XX and 88XXXXX pointing to this trunk.
Create an inbound route
Add an inbound route for the agreed numbers, pointing to an extension, ring group or queue.
Set up media
Set the codecs to alaw and ulaw and DTMF to rfc4733. Turn directmedia off, and turn rtpsymmetric and force_rport on if your PBX sits behind NAT.
Check the firewall
Allow SIP and RTP to and from our address.
Test
We check the connection, you call the agent's short number, and on request we call a transfer number with a test message.
Transferring to your team
- Cold: our platform calls your PBX with a second call.
- REFER (optional): Asterisk must allow transfers from this trunk. If Asterisk refuses, we fall back to the second call automatically.
- Warm: your colleague first hears a short summary, then presses 1 to take the call or 2 to decline it.
If nobody answers or the line is busy, the caller stays with the agent. The agent only calls numbers you have agreed with us in advance, and no call ever goes out to the public phone network over the trunk.
Points to watch
Related pages: connecting the AI agent to 3CX and to Yeastar. All platforms are listed in the integrations overview.
Connecting FreePBX or Asterisk?
Leave your details. We will call you back and send you the trunk details.
Asterisk and FreePBX are trademarks of Sangoma Technologies. Tibinet is not a partner of, or affiliated with, Sangoma.
Frequently asked questions
Yes, with a PJSIP trunk.
Yes. Point an IVR option to a Misc Destination with the agent's short number.
The summary arrives as a SIP header. Whether you display it depends on your own dial plan and handset.
An agreed maximum per connection. Above that we answer "busy".
Yes. Asterisk-based systems connect the same way, with a SIP trunk and two routes.
No. With registration your PBX signs in to us with a username and password. That also works behind NAT.
Call me back
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