SIP Trunking Getting Started

What Is SIP Trunking?

SIP trunking replaces traditional phone lines with an internet-based connection between your PBX (phone system) and the public telephone network. Benefits include:

• Lower cost - No physical phone lines. Pay per usage.
• Scalability - Add or remove channels instantly.
• Flexibility - Route calls based on time, location, or custom rules.
• Redundancy - Built-in failover across multiple regions.

AIVO Connect is AIVO's SIP trunking service, providing reliable voice connectivity for businesses of any size.

Creating Your First SIP Connection

1. Log in to your AIVO dashboard.
2. Navigate to AIVO Connect > SIP Trunking.
3. Click Create Connection.
4. Enter a name for the connection (e.g., "Office PBX" or "Call Center").
5. Choose your authentication method (see below).
6. Configure your settings and click Save.

Your connection is created immediately. You will see the SIP credentials and server addresses on the connection detail page.

Authentication Methods

AIVO Connect supports three authentication methods:

Credential Authentication

Best for most setups. Your PBX authenticates with a username and password.

1. Select Credential as the auth type.
2. AIVO generates a SIP username and password.
3. Enter these in your PBX trunk configuration.

IP Authentication

Best for static-IP environments. No username/password needed.

1. Select IP as the auth type.
2. Enter the public IP address of your PBX.
3. All SIP traffic from that IP is automatically authenticated.

Note: If your IP changes (e.g., dynamic ISP), use credential auth instead.

FQDN Authentication

For advanced setups with DNS-based routing.

1. Select FQDN as the auth type.
2. Enter your PBX's fully qualified domain name.
3. AIVO resolves the FQDN and authenticates based on the resolved IP.

Assigning Phone Numbers

1. Go to AIVO Connect > Numbers.
2. Click Get a Number to purchase a new number, or contact our team to port an existing number.
3. Select your country and area code.
4. Choose the number and click Assign to Connection.
5. Select the SIP connection you created.

Porting existing numbers: To port existing numbers to AIVO, contact our team at hello@aivo.bz. We'll handle the porting process for you.

Your number is now linked. Inbound calls to that number route to your PBX. Outbound calls from your PBX display that number as caller ID.

Making Your First Call

Inbound Test

1. Call your assigned AIVO number from a cell phone.
2. Your PBX should ring.
3. Answer the call and verify two-way audio.

Outbound Test

1. From a phone on your PBX, dial an external number.
2. Verify the call connects and you hear the other party.
3. Check that the correct caller ID is displayed.

Verifying in the Dashboard

After your test calls, go to AIVO Connect > Call Logs to confirm both calls appear with correct details.

Next Steps

• Configure FreePBX or Configure 3CX for step-by-step PBX setup.
• SIP Network Configuration for firewall and DNS setup.
• SIP Failover and Redundancy for high-availability setup.