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

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 [email protected]
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: [email protected])
2. In FlyNumber SIP Forwarding settings:
   • Domain: The provider's SIP server
   • Port: As specified by provider
   • Username: Your account/extension with the provider

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