Connect your PBX to a voice agent

You keep your PBX, your numbers and your extensions. The agent is one more SIP destination: the PBX sends it the calls you decide, all of them, only after hours, or only when nobody answers.

Step 1: declare the PBX

In Telephony › External PBXs add the PBX with a name, the customer it belongs to (or personal use) and the public IP address its calls come from. That is the whole authentication: no SIP registration, no password. If the address changes, update it before sending calls.

Step 2: dial the agent's SIP URI

The URI is in the agents list, like sip:ag_7f3a2b91@pbx.swissai.dev. Audio in G.711 alaw or ulaw.

; Asterisk, PJSIP: extension 200 rings the agent
exten => 200,1,Dial(PJSIP/ag_7f3a2b91@swissai,60)

<!-- FreeSWITCH dialplan -->
<action application="bridge" data="sofia/external/ag_7f3a2b91@pbx.swissai.dev"/>

# Kamailio / OpenSIPS
if ($rU == "200") { $ru = "sip:ag_7f3a2b91@pbx.swissai.dev:5060"; t_relay(); exit; }

Step 3: decide which calls go to the agent

• All calls on a number: the inbound route points to the agent.
• Time conditions: the PBX sends calls to the agent outside opening hours.
• Overflow: ring the team first, forward to the agent on no answer or busy.
• A dedicated extension, so staff can transfer a caller to the agent.

Transfers back to people

The agent has a built-in transfer tool: when the caller asks for a person, or the prompt says so, the call goes to the fallback number you set on the agent. That number can be an extension on your PBX.

Other PBXs

Any PBX or SBC that can dial a SIP URI to a host works the same way, including 3CX, FusionPBX and cloud PBXs that allow a SIP trunk without registration. If your PBX cannot send calls to an external SIP host, register the number on the platform instead: the platform registers with your operator and routes the calls to the agent.

Related pages

• A SIP URI for every voice agent
• After-hours and overflow: the calls nobody can take
• An AI receptionist that answers the phone