What is a SIP Connector?

A SIP Connector Server acts as the bridge between your existing PBX infrastructure and Mihu AI's voice agent platform. It translates SIP (Session Initiation Protocol) signaling so that incoming and outgoing calls from your PBX extensions can be routed to—and handled by—an AI voice agent.

Once connected, your PBX extensions behave as if they're dialing another internal extension, but on the other end, a fully capable AI agent answers, qualifies leads, schedules appointments, or handles support queries in real time.

Compatible PBX Systems

Mihu AI's SIP connector works with 3CX, Asterisk, FreePBX, and any standards-compliant SIP-based PBX. Both cloud-hosted and on-premise deployments are supported.

Prerequisites

Before you begin, make sure you have the following ready:

  • An active Mihu AI account with Voice Agent access
  • Admin access to your PBX system
  • The public IP address or FQDN of your PBX server
  • A designated port for SIP traffic (default: 5060 for UDP/TCP, 5061 for TLS)
  • At least one available PBX extension to assign to the voice agent

Step 1 — Create the SIP Connector Server

Navigate to the Connectors section in your Mihu AI dashboard and create a new SIP Connector Server. This registers a connection endpoint that your PBX will send calls to.

Open the SIP Connector panel

From your Mihu AI dashboard, go to Settings → Connectors → SIP and click "New SIP Server". Give it a descriptive name that identifies the PBX it connects to, such as "Office-3CX-Main" or "Clinic-Asterisk-PBX".

Enter the server connection details

Fill in the server hostname or IP, the SIP port, and select the transport protocol. TLS is recommended for production environments to ensure encrypted call signaling.

NAT Considerations

If your PBX sits behind a NAT firewall, use the public IP address or FQDN here—not the private/internal IP. Make sure ports 5060-5061 and your RTP media range (e.g. 10000-20000) are forwarded correctly.

Step 2 — Add a SIP Trunk

With the server created, you now need to add a SIP trunk—this defines the specific authentication credentials and routing rules that link your PBX extensions to the voice agent.

Create a new trunk on the server

Inside the SIP Server you just created, click "Add Trunk". Each trunk represents a logical connection between a set of PBX extensions and a Mihu AI voice agent.

Set authentication credentials

Enter the SIP username and password. These must match the credentials configured on your PBX for the outbound trunk. Use a strong, unique password for security.

Step 3 — Configure Trunk Details

Here's a reference table of the key fields you'll need to configure when setting up your SIP trunk:

Field Description Example
Trunk Name A friendly label for identifying this trunk Office-Main-Trunk
SIP Username The authentication username for the trunk mihu_trunk_01
SIP Password Secure password matching your PBX config ••••••••
Registrar The address the trunk registers with sip.mihu.ai
Transport Protocol for SIP signaling TLS (recommended)

Real-World Use Cases

Different industries use SIP trunks with Mihu AI in different ways. Here are four common scenarios showing how businesses route calls through their PBX to an AI agent:

Healthcare: After-Hours Triage

Route all calls outside business hours to an AI agent that collects patient symptoms, schedules urgent callbacks, and handles prescription refill requests.

Routing rule:
Time condition: 6PM–8AM → Extension 900 (Mihu AI)
Fallback: Voicemail after 3 rings

Automotive: Service Department Overflow

When all service advisors are busy, overflow calls go to AI that books service appointments, provides repair status updates, and quotes common services.

Routing rule:
Ring group 200 (Service) → No answer 20s → Extension 901 (Mihu AI)

Real Estate: Lead Qualification

Incoming leads from property listings hit AI first to qualify budget, timeline, and preferences before routing hot leads directly to available agents.

Routing rule:
DID +1-555-HOMES → Extension 902 (Mihu AI)
Qualified lead → Transfer to Ring Group 300

E-Commerce: Order Status & Returns

Handle high-volume order inquiries with AI that looks up tracking info, processes return requests, and escalates complex issues to human support.

Routing rule:
IVR option 1 (Order Status) → Extension 903 (Mihu AI)
IVR option 2 (Returns) → Extension 904 (Mihu AI)

PBX Routing Scenarios

Here's how to configure common routing scenarios in your PBX to work with Mihu AI:

PBX Configuration Examples

Copy these patterns into your PBX dial plan or routing rules

1

Direct Extension Routing

The simplest setup: assign a dedicated extension to the AI agent. Any call to this extension goes directly to Mihu AI. Ideal for testing or dedicated AI lines.

; 3CX / Asterisk dial plan exten => 900,1,Dial(SIP/mihu-trunk/900,30) exten => 900,n,Hangup() ; FreePBX: Create extension 900, set destination to "Custom Trunk: mihu-trunk"
2

Time-Based Routing (After Hours)

Route calls to AI only during specific hours. During business hours, calls go to human agents; after hours, they go to the AI agent for 24/7 coverage.

; Time condition: Mon-Fri 9AM-6PM → Ring Group, else → AI exten => s,1,GotoIfTime(09:00-18:00,mon-fri,*,*?business:afterhours) exten => s,n(business),Dial(SIP/ring-group-100,30) exten => s,n,Goto(afterhours) exten => s,n(afterhours),Dial(SIP/mihu-trunk/900,30) exten => s,n,Hangup()
3

Overflow Routing (All Agents Busy)

When all human agents are on calls, overflow to AI instead of putting customers on hold. The AI handles the inquiry or takes a message for callback.

; Ring group with overflow to AI after 20 seconds exten => 100,1,Dial(SIP/agent1&SIP/agent2&SIP/agent3,20) exten => 100,n,Dial(SIP/mihu-trunk/900,30) exten => 100,n,Hangup() ; FreePBX: Ring Group → Destination if no answer: "Custom: mihu-trunk/900"
4

IVR Menu Integration

Add AI as an option in your IVR menu. Customers can choose to speak with AI for quick tasks (order status, appointments) or wait for a human agent.

; IVR: "Press 1 for AI assistant, Press 2 for human agent" exten => 1,1,Dial(SIP/mihu-trunk/900,30) exten => 1,n,Hangup() exten => 2,1,Dial(SIP/ring-group-sales,60) exten => 2,n,VoiceMail(sales@default) exten => 2,n,Hangup()
5

Caller ID-Based Routing (VIP vs. General)

Route callers based on their phone number. VIP customers go directly to human agents; general inquiries go to AI for initial handling.

; Check caller ID against VIP list exten => s,1,GotoIf($[${DB_EXISTS(vip/${CALLERID(num)})}]?vip:general) exten => s,n(vip),Dial(SIP/priority-queue,30) exten => s,n,Hangup() exten => s,n(general),Dial(SIP/mihu-trunk/900,30) exten => s,n,Hangup()

Pro Tip: Warm Transfers

Configure your AI agent to perform warm transfers back to human agents. The AI can brief the human on what was discussed before connecting the caller, saving time and improving customer experience.

Step 4 — Verify and Test

After saving the trunk configuration, verify the connection status in your Mihu AI dashboard. The trunk status should show as "Registered" within 30 seconds.

Check registration status

Go back to Settings → Connectors → SIP and confirm the trunk shows a green "Registered" badge. If it shows "Failed" or "Timeout", double-check your credentials and network configuration.

Make a test call

From any PBX extension, dial the extension number assigned to the Mihu AI voice agent. You should hear the AI agent greeting within 2-3 seconds. If the call connects but there's no audio, check your RTP port forwarding.

Troubleshooting

If you run into issues, here are the most common problems and their solutions:

Registration fails

Verify that the SIP username, password, and registrar address are correct. Check that your firewall allows outbound SIP traffic on port 5060 (UDP/TCP) or 5061 (TLS). If your PBX is behind NAT, ensure the public IP is configured correctly in both the PBX and Mihu AI settings.

One-way or no audio

This is almost always a NAT or RTP issue. Ensure RTP ports (typically 10000–20000) are forwarded through your firewall. Enable STUN/TURN if your PBX supports it for better NAT traversal. Also verify that both sides agree on audio codecs—G.711 μ-law (PCMU) is the safest choice for compatibility.

Calls drop after 30 seconds

This usually indicates a SIP session timer mismatch. Check that your PBX's session timer settings are compatible. Some PBX systems send re-INVITE or UPDATE messages that need to be handled correctly—make sure these aren't being blocked by your firewall.

Need Help?

If you're still having trouble, contact Mihu AI support with your trunk ID and a SIP trace log from your PBX. This helps the team diagnose the issue quickly.