SIP Forwarding

SIP Forwarding connects your FlyNumber directly to an external SIP address or URI—like your own PBX, a VoIP provider, or any SIP-enabled system. Unlike PSTN forwarding which sends calls to regular phone numbers, SIP forwarding keeps everything digital, often resulting in better audio quality and more flexibility.

When to Use SIP Forwarding

Choose SIP forwarding when you:

• Already have a PBX system (Asterisk, FreePBX, 3CX, etc.) and want calls to ring there
• Use a third-party VoIP provider for call handling
• Need to connect multiple phone systems together
• Want to avoid per-minute PSTN forwarding charges

Routing Options

See Routing Options for two ways to configure how calls reach your SIP system—either direct assignment or through call flows.

Compatible Systems

SIP forwarding works with any SIP-compliant system. Here are some popular options:

System	Type	Notes
Asterisk	Open-source PBX	Most flexible, requires technical setup
FreePBX	GUI for Asterisk	Easier to manage than raw Asterisk
3CX	Commercial PBX	Available as cloud or self-hosted
FreeSwitch	Open-source softswitch	High-performance, scalable
VoIP Providers	Varies	Check with provider for SIP URI format
AI Voice Platforms	VAPI, LiveKit, ElevenLabs, Retell	See transport requirements below

Setting Up SIP Forwarding

Step 1: Gather Your SIP Details

Before configuring, you'll need the following from your SIP system or provider:

• SIP URI or domain (e.g., sip.yourprovider.com or an IP address)
• Port number (typically 5060 for UDP/TCP, 5061 for TLS)
• Username (if authentication is required)
• Any firewall/security requirements

Step 2: Create SIP Forwarding Contact Method

1. Navigate to Contact Methods in the main menu
2. Select the SIP Forwarding tab
3. Click the + button to create a new configuration

Step 3: Configure General Settings

Setting	Description	Example
User	Select the user who will own this configuration	John Smith
Username	Your SIP username or extension	1001 or john@sip.example.com
Domain	The SIP domain or IP address	sip.yourprovider.com or 192.168.1.100
Port	Port number for SIP communication	5060 (standard), 5061 (TLS)
Network Protocol	IP version preference	IPv4 Only (most common)
Process 30X redirects	Handle SIP redirects automatically	Enable if your system uses redirects

Network Protocol Options

Option	When to Use
IPv4 Only	Most common—use this if unsure
IPv6 Only	When your system only supports IPv6
Dualstack	System supports both IPv4 and IPv6
IPv4 Preferred	Try IPv4 first, fall back to IPv6
IPv6 Preferred	Try IPv6 first, fall back to IPv4

IP Whitelisting

If your SIP system uses IP-based security (firewall rules, ACLs), you'll need to whitelist our IP ranges to receive calls.

FlyNumber IP Ranges

For SIP signaling (incoming and outgoing traffic):

46.19.208.0/21 (46.19.208.0 - 46.19.215.254)

For RTP media (voice audio):

Ports: 16384-32767

Firewall Configuration

If you're running your own PBX behind a firewall, make sure both the SIP ports AND the RTP port range are open. Missing RTP ports is a common cause of one-way audio issues.

Example: Adding to iptables (Linux)

# Allow SIP signaling from FlyNumber
iptables -A INPUT -s 46.19.208.0/21 -p udp --dport 5060 -j ACCEPT

# Allow RTP media
iptables -A INPUT -s 46.19.208.0/21 -p udp --dport 16384:32767 -j ACCEPT

Common Setup Examples

Asterisk / FreePBX

1. Create an inbound route in your Asterisk/FreePBX system
2. Note your server's public IP or domain
3. In FlyNumber SIP Forwarding settings:
   • Domain: Your server's public IP or domain
   • Port: 5060 (or your configured SIP port)
   • Username: The extension or context you want calls routed to
4. Whitelist FlyNumber IPs in your Asterisk firewall

3CX

1. In 3CX, go to SIP Trunks and configure an inbound trunk
2. Note the trunk's registration details
3. In FlyNumber SIP Forwarding settings:
   • Domain: Your 3CX FQDN or IP
   • Port: As specified by 3CX (usually 5060)
   • Username: Your SIP trunk username
4. Configure inbound routing in 3CX to direct calls appropriately

External VoIP Provider

1. Get your SIP URI from your provider (often in format: username@sip.provider.com)
2. In FlyNumber SIP Forwarding settings:
   • Domain: The provider's SIP server
   • Port: As specified by provider
   • Username: Your account/extension with the provider

AI Voice Platforms

AI voice agents are increasingly used for automated customer service, appointment scheduling, and 24/7 call handling. Here's what to know when connecting to these platforms:

Platform	Transport	Notes
VAPI	UDP, TCP, TLS	UDP is default; see VAPI example below
LiveKit	UDP, TCP, TLS	Open-source agent runtime; see LiveKit example below
ElevenLabs	TLS required	UDP not supported; TLS is a requirement
Retell AI	TCP, TLS	Check their current documentation
Bland AI	TCP, TLS	Check their current documentation

Setup steps:

1. Get the SIP URI from your AI platform
2. Configure SIP Forwarding with the platform's domain and your credentials
3. Select the appropriate transport protocol (TCP can be set in the panel; TLS requires support)
4. Whitelist FlyNumber IPs on the AI platform if required

Transport Protocol

You can change the transport protocol (UDP/TCP) directly in the phone system panel. For AI platforms like ElevenLabs that require TLS, contact support to enable TLS for your account.

VAPI Example

VAPI assigns each SIP trunk a unique credential ID. The SIP URI follows this pattern:

+{your_flynumber}@{credential_id}.sip.vapi.ai

To set this up in the phone system panel:

1. In your VAPI dashboard, create a BYO SIP Trunk (choose the "Bring Your Own" option, not a standard credential)
2. Add all 5 FlyNumber gateway IPs as separate gateways in the trunk configuration:
   • 46.19.209.14
   • 46.19.210.14
   • 46.19.212.14
   • 46.19.213.14
   • 46.19.214.14
3. For each gateway, enable Inbound Calls and Options Ping — this allows VAPI to accept and monitor incoming calls from FlyNumber
4. VAPI will generate a credential ID (a UUID like a1b2c3d4-5678-90ab-cdef-1234567890ab)
5. Register your FlyNumber as a phone number in VAPI and associate it with your AI assistant
6. In the SIP Forwarding contact method, configure:

Setting	Value
Domain	{credential_id}.sip.vapi.ai (use your actual credential ID)
Username	Your FlyNumber in E.164 format (e.g., +13025551234)
Port	5060

7. VAPI supports UDP (default), TCP, and TLS — select TCP directly in the panel, or contact support for TLS

BYO SIP Trunk Required

A standard VAPI SIP trunk credential won't work with FlyNumber. You must use the BYO (Bring Your Own) SIP Trunk option and add all FlyNumber gateway IPs as gateways with inbound calls enabled. Without this, calls will return "Service Unavailable."

VAPI does not require SIP registration — the credential ID in the domain handles authentication.

LiveKit Example

LiveKit is an open-source realtime platform often used to host voice AI agents. Each LiveKit Cloud project has its own SIP URI — something like abcd1234.sip.livekit.cloud — visible on the Project settings page of the LiveKit dashboard. Calls from your FlyNumber are delivered into a LiveKit room where your agent logic picks them up.

Three pieces have to be in place on LiveKit before the call completes: an inbound trunk containing your DID, a dispatch rule that sends matching calls into a room, and an agent subscribed to that room. Missing any one will still surface as a failed call on the FlyNumber side.

Step 1 — Copy your SIP URI

In the LiveKit Cloud dashboard, open Project settings and copy the SIP URI. Drop the sip: prefix — what's left is your project's SIP endpoint (for example, abcd1234.sip.livekit.cloud).

Step 2 — Create the inbound trunk

Go to Telephony → SIP trunks → Create new trunk. On the Trunk details tab, set Trunk direction to Inbound, type your FlyNumber in E.164 into the Numbers field (comma-separated if you're attaching multiple DIDs), and leave Allowed addresses blank unless you want to lock the trunk to specific IPs.

Figure: Inbound trunk configuration in the LiveKit Cloud dashboard. The Numbers field has to include the DID the call is arriving for.

Prefer the JSON editor?

Switch to the JSON editor tab on the same dialog and paste:

{
  "trunk": {
    "name": "FlyNumber inbound",
    "numbers": ["+17185301799"]
  }
}

If numbers is empty and no allowedAddresses or auth credentials are set, LiveKit rejects the trunk — a wildcard trunk always needs one of those guard rails.

Step 3 — Create the dispatch rule

Under Telephony → Dispatch rules → Create new dispatch rule, pick Individual as the rule type and set a room prefix such as call-. Each call lands in a freshly generated room with that prefix. Leave Inbound routing at Phone numbers / all trunks unless you want to scope the rule.

Figure: Dispatch rule creation. Without a published dispatch rule, LiveKit accepts the INVITE on the trunk and then drops the call because no room has been assigned.

Dispatch rule via JSON

{
  "name": "FlyNumber dispatch",
  "rule": {
    "dispatchRuleIndividual": {
      "roomPrefix": "call-"
    }
  }
}

Step 4 — Configure the SIP Forwarding contact method

In the phone system panel, set the SIP Forwarding fields to:

Setting	Value
Domain	{project}.sip.livekit.cloud (the SIP URI from your Project settings)
Username	Your FlyNumber in E.164 format (e.g., +17185301799)
Port	5060

LiveKit listens on UDP/5060 by default — leave the default transport alone unless you've turned on secure trunking, in which case switch the panel to TCP/TLS on port 5061 and contact support if TLS still needs to be enabled on your account.

Step 5 — Attach an agent

Publishing the trunk and dispatch rule gets the call into a LiveKit room, but nothing happens until an agent subscribes to that room and answers. Make sure your agent (custom or deployed via LiveKit Agents) is configured to handle rooms created by your dispatch rule before testing.

Username must be the DID in E.164

The LiveKit inbound trunk routes calls by matching the To: user in the SIP INVITE against the numbers array on the trunk. If the username field holds a friendly name — say, agent1 — instead of the DID in E.164 (for example, +17185301799), LiveKit returns 404 Not Found and your FlyNumber call log shows status fail, reason not found. The call won't appear in the LiveKit dashboard at all, because the rejection happens at the trunk layer before any dispatch rule runs.

Trunk + dispatch rule are both required

The inbound trunk accepts the call. The dispatch rule places it into a room. If either is missing, the call fails — publish both before testing.

IP whitelisting

LiveKit only enforces IP restrictions when allowedAddresses is set on the inbound trunk. If you've added that restriction, include the FlyNumber IP range from the IP Whitelisting section above.

Troubleshooting

Problem	Likely Cause	Solution
Calls don't arrive	Firewall blocking	Whitelist FlyNumber IPs
One-way audio	RTP ports blocked	Open UDP ports 16384-32767
Call connects then drops	Authentication failure	Verify username and domain
Intermittent failures	NAT issues	Check NAT settings on your PBX
"404 Not Found" errors	Invalid username/extension	Verify the destination exists on your system

Testing Your Setup

1. Check connectivity: Ensure your SIP system is reachable from the internet
2. Test with a simple call: Make a test call to your FlyNumber
3. Check logs: Review SIP logs on your PBX for connection attempts
4. Verify audio path: Confirm two-way audio works

Debugging Tip

Most PBX systems have SIP debug logging. Enable this temporarily to see exactly what's happening when calls arrive. Look for INVITE messages from our IP range.

SIP Forwarding vs PSTN Forwarding

Feature	SIP Forwarding	PSTN Forwarding
Destination	SIP URI / IP address	Regular phone number
Audio Quality	Typically higher (digital end-to-end)	Good (converted to PSTN)
Per-minute cost	No additional charges	Per-minute rates apply
Setup complexity	Requires SIP knowledge	Simple (just enter phone number)
Best for	PBX systems, VoIP providers	Mobile phones, landlines

note

This is different from traditional call forwarding—instead of forwarding calls through the phone network, you're routing them directly to your SIP-enabled system. Check out our blog post on Forwarding Calls vs Setting Calls.

Best Practices

1. Security: Always use IP whitelisting or authentication on your PBX
2. Redundancy: Consider a backup contact method in case your SIP system is unreachable
3. Monitoring: Set up alerts for when your PBX goes offline
4. Documentation: Keep a record of your SIP configuration for troubleshooting
5. Testing: After setup, test calls from different sources to ensure reliability