SIP Trunking Documentation

AIVO Connect provides enterprise SIP trunking that connects your PBX, softphones, or communication platform to the global telephone network. This guide will walk you through setup from start to finish.

The setup process takes about 5 minutes: create a connection, assign numbers, and point your PBX to our SIP domain.

AIVO Connect SIP trunking is available across multiple plan tiers to fit businesses of every size.

Pay-As-You-Go (PAYG)

The PAYG plan has no monthly fee. You pay only for the minutes you use, billed at per-destination rates with 6-second billing increments rounded in your favour and a 1-cent minimum per connected call.

• US Inbound: $0.008 BZD/min • US Outbound: $0.012 BZD/min
• Belize Inbound: $0.015 BZD/min • Belize Outbound: $0.020 BZD/min
• 19 supported destinations — see the full rate table at /connect/rates
• 1 free phone number included; additional local numbers $3 BZD/mo, toll-free $6 BZD/mo
• $2 BZD minimum balance required for outbound calls
• Daily spend limit: $50 BZD/day ($10 BZD/day for the first 7 days)

Subscription Plans

For more connections, phone numbers, and priority support, AIVO Connect offers subscription tiers.

You can start with PAYG and upgrade to a subscription plan at any time from your dashboard. All plans use the same SIP infrastructure and setup process described below.

A SIP connection defines how your equipment authenticates with AIVO Connect. Go to SIP Trunking > Connections and click "Create Connection."

1. Choose an authentication type (Credential, IP, or FQDN)
2. Enter a descriptive connection name
3. Fill in the authentication details for your chosen method
4. Optionally select an anchorsite override for geographic preference
5. Click "Create Connection"

Once created, you will see your connection credentials (for credential auth) or connection ID (for all types) in the connection detail view.

Credential Authentication

The most common method. Your PBX registers with a username and password, similar to a SIP phone registration. This works behind NAT and firewalls without special configuration.

IP Authentication

For equipment with static public IPs. No registration is needed; calls from whitelisted IPs are automatically accepted. Ideal for data center deployments.

FQDN Authentication

Authenticate using a fully qualified domain name. Your PBX presents its FQDN during SIP registration, and AIVO Connect verifies it via DNS.

Phone numbers must be assigned to a SIP connection before they can receive or place calls through your trunk.

1. Purchase numbers from Settings > Numbers
2. Go to SIP Trunking > Numbers
3. Click the link icon next to an unassigned number
4. Select the SIP connection to assign it to

Numbers can be reassigned between connections at any time. Changes take effect within seconds.

Outbound voice profiles control how your outbound calls are routed and billed. You can set daily spend limits to prevent unexpected charges.

• Name: A descriptive label for the profile
• Daily Spend Limit: Maximum USD amount per day (optional)
• Enabled: Toggle the profile on/off without deleting it

Each SIP connection can be associated with one outbound voice profile. If a daily spend limit is reached, outbound calls from that profile will be rejected until the next day.

SIP Signaling

Media (RTP)

Media ports range from UDP 20000-60000. If you have a firewall, ensure these port ranges are open for the AIVO Connect IP ranges.

Supported Codecs

• G.711 u-law (PCMU)
• G.711 a-law (PCMA)
• G.729
• Opus
• T.38 (fax)

DTMF

RFC 2833 is the recommended DTMF method. SIP INFO and in-band DTMF are also supported.

AIVO Connect automatically handles failover at multiple levels:

• DNS-based failover: Multiple A records for sip.aivo.bz ensure your PBX can reach an alternate server if one is unavailable.
• Anchorsite failover: If your selected anchorsite experiences issues, traffic is automatically rerouted to the nearest healthy PoP.
• Carrier failover: Outbound calls automatically retry on alternate carrier routes if the primary route fails.

For maximum reliability, configure your PBX with multiple SIP servers (primary and backup) pointing to our SIP domain.

All SIP trunking operations are available via the AIVO REST API. Authenticate with your API key using a Bearer token.

Create a Credential Connection

curl -X POST https://aivo.bz/api/v1/sip/connections \
  -H "Authorization: Bearer aivo_your_api_key" \
  -H "Content-Type: application/json" \
  -d '{
    "auth_type": "credential",
    "connection_name": "Office PBX",
    "user_name": "office-pbx",
    "password": "your-secure-password",
    "anchorsite_override": "Miami"
  }'

List Connections

curl https://aivo.bz/api/v1/sip/connections \
  -H "Authorization: Bearer aivo_your_api_key"

Create Outbound Voice Profile

curl -X POST https://aivo.bz/api/v1/sip/outbound-profiles \
  -H "Authorization: Bearer aivo_your_api_key" \
  -H "Content-Type: application/json" \
  -d '{
    "name": "Main Outbound",
    "daily_spend_limit": "100.00",
    "enabled": true
  }'

Delete a Connection

curl -X DELETE "https://aivo.bz/api/v1/sip/connections/CONNECTION_ID?type=credential" \
  -H "Authorization: Bearer aivo_your_api_key"