Technical details for the SIP trunk

This page is for the IT team or installer setting up the trunk to Tibinet's AI phone agent. It covers the connection methods, our settings, numbering and transfers, and the checklist. Anything that differs per platform is on the page for your phone system.

Technical details for the SIP trunk

Two ways to connect

IP-based connectionRegistration
How we recognise your systemBy the public IP address of your phone system or SBCBy username and password
What you needA fixed public IP address (or a small block, a /24 at most)Nothing special, it also works behind NAT
Where we send callsThe address and port you give usThe address your system registers from
Usually chosen forMitel, Avaya, Cisco CUBE, Teams and Zoom SBCs, AudioCodes, Ribbon3CX, FreePBX, Yeastar, Grandstream, smaller phone systems

Settings on our side

Address
The SIP address we give you
Port
5060 (UDP and TCP), 5061 (TLS)
Transport
UDP, TCP or TLS, as agreed
Media (RTP)
UDP ports 10000 to 20000 inclusive, both directions
Codecs
G.711 A-law (preferred) and G.711 µ-law; G.722 (HD voice) on request
DTMF (keypad tones)
RFC 4733 (also known as RFC 2833, "telephone-event"); no SIP INFO, no in-band
Encryption
SRTP (SDES), only together with TLS; optional or mandatory
Reachability
We send a SIP OPTIONS to your phone system every 60 seconds
Concurrent calls
An agreed maximum per connection, both directions combined; above that we respond with "486 Busy Here"

Numbering

From your phone system to the agent. Every agent has a fixed seven-digit number starting with 88. We can also set up a short number per agent that fits your dial plan, for example 8500 for the AI reception desk. Your phone system sends the call with that number to the trunk. The caller's number may be included in the From header; the agent can use it. That means an IVR menu, a queue overflow or an out-of-hours rule on your phone system can simply point to the agent.

From the agent back to your phone system. The agent only calls numbers on your phone system that you agree with us in advance, for example 200 (reception) or 210 (sales queue). It will not call a number that is not on that list, even if the caller asks. No call ever goes out to the public telephone network over this trunk.

Transfers

  • Cold transfer. By default our platform places a second call (a second call leg) to your phone system. This works with any phone system. For as long as the call lasts, it uses two channels on the trunk.
  • Transfer with SIP REFER (optional). If the caller reached the agent through your own phone system, we can ask your system to transfer the call itself (blind transfer). Our part of the call then ends, which saves channels. Your phone system must accept REFER from this trunk. If it rejects the REFER or the transfer fails, the platform automatically falls back to the second call. REFER is off by default.
  • Warm transfer. The caller stays with the agent, with hold music if you like. Your colleague first hears a short summary of the call and then presses 1 (accept, the caller is put straight through) or 2 (decline, the caller stays with the agent and notices nothing). No key press, no answer or a busy line counts as declining.
  • Fallback. If nobody answers or the line is busy during a cold transfer, the caller goes back to the agent.

With a warm transfer, the summary is also sent as a SIP header with the call to your phone system: a single line of up to 200 characters. If your phone system or handsets can display SIP headers, your colleague will see it on screen. Whether that is possible depends on your system and handsets; we do not promise it. An SBC may also strip such a header. The connection works without it.

Checklist for your IT team

  1. Create the trunk

    A SIP trunk (or SIP peer, SIP line, external trunk) to the address we give you, using the connection method you chose.

  2. Send us your details

    For an IP-based connection: send us the public IP address, port and transport of your phone system or SBC.

  3. Firewall and SBC

    Allow SIP (5060 or 5061) and RTP (10000 to 20000) to and from our address.

  4. Outbound route

    Create an outbound route in the dial plan for the agent numbers (for example 85XX and 88XXXXX) to the trunk.

  5. Inbound route

    Create an inbound route for the transfer numbers you agree with us, so they arrive as internal calls.

  6. Codec and DTMF

    Enable the G.711 A-law codec and set DTMF to RFC 4733.

  7. Optional

    TLS and SRTP, and allowing REFER (blind transfer) on this trunk.

  8. Test

    We send OPTIONS, you call an agent's short number, and on request we call a transfer number with a test message.

How long this takes depends on your phone system, your SBC and your change process. The overview for business owners is on the page connect an AI phone agent to your phone system. No SIP trunk possible? Read about forwarding calls to the agent's number.

Need the trunk details?

Tell us which phone system or SBC you use and which connection method you prefer. We will call you back and send the SIP address and the other details.

Frequently asked questions

G.711 A-law and µ-law. G.722 (HD voice) on request.

According to RFC 4733, also known as RFC 2833 or "telephone-event". We do not use SIP INFO or in-band DTMF.

It can be. The trunk runs over TLS, with SRTP for the audio. You choose whether encryption is optional or mandatory.

We agree that per connection. Above the maximum, our platform responds with "busy" (486), so your phone system can choose another route.

Yes. Your phone system then registers with us using a username and password. That also works behind NAT.

With a cold transfer using a second call, two channels on the trunk for as long as the call lasts. With REFER, our part of the call ends and you save channels.

No. The agent only calls the numbers on your phone system that you agree with us in advance. No call ever goes out to the public telephone network over the trunk.

Call me back

Leave your name and number. We will call you back at a time that suits you.