Set to VoIP
Configure your FlyNumber to forward calls directly to VoIP. This allows you to integrate with any VoIP provider or PBX system while maintaining unlimited incoming minutes.
Configuration Steps​
- Log in to your FlyNumber account at My Account.
- Navigate to My FlyNumbers.
- Find the FlyNumber you want to configure and click Details.
- Click Change forwarding to access the forwarding options.
- Select VoIP from the forwarding options.
Figure 1: Configuration options for setting the FlyNumber directly to VoIP from the My FyNumbers page.
Figure 2: Configuration options for setting the FlyNumber directly to VoIP from the Signup/ Add FlyNumber page.
Understanding VoIP Fields​
When setting up VoIP forwarding, you'll need to configure three main fields:
1. Protocol​
- Default: SIP (Session Initiation Protocol)
- Options:
- SIP: Most common protocol for VoIP communications
- H.323: Legacy protocol for real-time audio/video communication
- IAX2: Inter-Asterisk eXchange protocol, primarily used with Asterisk systems
- Best Practice: Use SIP unless your system specifically requires another protocol
2. Host​
- For VoIP Providers:
- The part after @ in your SIP address
- Example: If your SIP address is [email protected], enter
sip.acme.com
- For PBX Systems:
- Your server's public IP address or hostname
- Example: If your Asterisk server is at 12.34.45.7, enter
12.34.45.7
3. Username/extension​
- For VoIP Providers:
- The part before @ in your SIP address
- Example: If your SIP address is [email protected], enter
john
- For PBX Systems:
- Your extension number (e.g.,
101) - The username configured in your PBX
- Your extension number (e.g.,
Technical Specifications​
IP Addresses​
Calls originate from the following endpoints:
- 46.19.209.14:5060 (New York POP)
- 46.19.210.14:5060 (Frankfurt POP)
- 46.19.212.14:5060 (Los Angeles POP)
- 46.19.213.14:5060 (Miami POP)
- 46.19.214.14:5060 (Singapore POP)
RTP traffic subnet:
- 46.19.208.0/21
- Port range: 1024-65535
Supported Codecs​
- G.711 A-law/U-law
- G.729
- G.723.1
- L16
- G.726-16/G.726-40/G.726-32/G.726-24
- G.721
- GSM
- Speex
DTMF Transport​
- Telephone-event RFC2833 (default)
- SIP INFO draft-kaplan-dispatch-info-dtmf-package-00
RTCP Support​
- Transmits and receives on port = rtp_port + 1 (RFC3550)
- Supports RTCP conflict avoidance payloads (72-76)
Popular Integration Examples​
PBX Systems​
- Asterisk: Configure extensions and SIP trunks
- FreePBX: Set up inbound routes and trunks
- 3CX: Create SIP trunk with FlyNumber credentials
- FreeSWITCH: Set up SIP profiles and dialplans
VoIP Providers​
- Twilio: Forward to SIP domains
- Callcentric: Use SIP URI forwarding
- Telnyx: Configure SIP trunking
- Flowroute: Set up inbound routes
- OnSIP: Use inbound bridge configuration
Set "host" to iptel.org and "user/extension" to music to quickly test your FlyNumber. You'll hear music when you call the number.
Troubleshooting​
Common Call Responses​
Check your call logs for these responses:
- Time Out: No response from VoIP server
- Proxy Authentication Error: Check IP whitelist settings
- Forbidden: Registration required or configuration issue
- OK: Call successfully connected
Best Practices​
- Whitelist FlyNumber IP addresses in your system's ACL
- Verify your VoIP system accepts incoming calls
- Test configuration with a test call
- Monitor call logs for any issues
- Ensure proper codec support
If using systems like Asterisk or FreePBX, you may need to whitelist our IP addresses in your system's Access Control List (ACL).
What's Next?​
After configuring VoIP forwarding:
- Make a test call to your FlyNumber
- Check your call logs for successful connection
- Verify audio quality and connection stability
- Configure any additional features in your VoIP system
For advanced call handling features like auto-attendants, time-based routing, or call recording, consider using our cloud phone system.