Fixed Line & SIP Trunk
Set up fixed-line numbers and SIP trunk connections to power AI calls on Fonify.
Fixed Line & SIP Trunk
SIP trunking lets you connect your existing telephony infrastructure to Fonify over the internet. This is the most cost-effective option for high-volume calling and works with fixed-line numbers from any provider.
What is SIP Trunking?
SIP (Session Initiation Protocol) trunking replaces traditional phone lines with a virtual connection over IP. Instead of physical copper wires, voice traffic flows through your internet connection to Fonify's call engine.
Cost Savings
Eliminate per-line fees. Pay only for concurrent call channels you actually use.
Scalability
Add or remove channels instantly without waiting for physical line installation.
Number Flexibility
Keep your existing numbers or add new ones. Port numbers between providers easily.
Global Reach
Route calls internationally without separate international calling plans.
Prerequisites
Before setting up a SIP trunk on Fonify, ensure you have:
- A SIP trunk account with a telephony provider (e.g., Turk Telekom, Twilio, Telnyx, VoIP.ms)
- SIP credentials: server address, port, username, and password
- A static IP address (if your provider uses IP-based authentication)
- Sufficient internet bandwidth (approximately 100 Kbps per concurrent call)
Fonify supports both registration-based and IP-based SIP authentication. Check which method your provider uses before starting.
Setting Up a SIP Trunk
Gather the SIP details you received from your provider and share them with support to start the Fonify-side setup.
The support team will need the connection details provided by your SIP provider:
- SIP Server: The hostname or IP address (e.g.,
sip.provider.com) - Port: Usually 5060 (UDP) or 5061 (TLS)
- Username: Your SIP authentication username
- Password: Your SIP authentication password
- Transport: Select UDP, TCP, or TLS
Fonify supports G.711 (PCMU/PCMA), G.729, and Opus codecs. Select the codecs your provider supports. G.711 offers the best quality; G.729 uses less bandwidth.
Confirm which phone number(s) should be used as caller ID on outbound calls.
Run a verification call with support to confirm SIP registration and audio quality.
Fixed-Line Number Setup
Fixed-line numbers are geographic numbers tied to a specific area code. To use a fixed-line number with Fonify:
Request SIP access for your existing fixed-line number. Providers like Turk Telekom offer this through their business divisions.
Depending on your provider, you either forward calls to a Fonify SIP endpoint or register the number directly through SIP credentials.
Share the credentials your fixed-line provider gave you with support so the SIP trunk can be connected for you.
Not all fixed-line providers support SIP access on residential lines. You may need a business-grade line or a separate SIP trunk service.
Troubleshooting
| Issue | Likely Cause | Solution |
|---|---|---|
| Registration failed | Wrong credentials or firewall blocking | Verify credentials; open UDP port 5060 |
| One-way audio | NAT/firewall blocking RTP | Configure STUN/TURN or open RTP port range (10000-20000) |
| Call drops after 30s | SIP session timer mismatch | Check provider's session timer settings |
| Poor audio quality | Insufficient bandwidth or codec mismatch | Switch to G.711; ensure 100 Kbps per call |
| Caller ID not showing | Outbound CLID not configured | Set the correct number in Fonify's caller ID field |
Tip
Enable TLS (port 5061) whenever your provider supports it. This encrypts SIP signaling and prevents eavesdropping on call metadata.
Advanced Configuration
- Failover routing: Configure a secondary SIP trunk as a backup. If the primary fails, Fonify automatically routes calls through the secondary.
- Concurrent call limits: Set the maximum number of simultaneous calls to match your SIP trunk capacity.
- DTMF mode: Choose between RFC 2833 (out-of-band) and in-band DTMF depending on your provider's requirements.