How to Configure SIP Carriers in VICIdial

Why SIP Carriers Are the Foundation of Your Dialer

Every call your vicidial dialer places or receives travels through a SIP carrier. Without a properly configured SIP trunk, your vicidial server can manage campaigns, load leads, and log in agents—but no audio will ever reach a phone. SIP carrier configuration is the single most important telephony step in your vicidial setup, and it is the step most likely to cause headaches if rushed.

This guide covers the complete process of connecting VICIdial to a SIP provider—from gathering credentials and creating carrier records to configuring dial prefixes, outbound routes, and failover. Whether you use a budget SIP provider or an enterprise-grade carrier, the configuration workflow in VICIdial follows the same pattern.

Gathering Carrier Credentials

Before opening the VICIdial admin panel, collect the following information from your SIP provider. Most carriers deliver these details in a welcome email or customer portal after account activation.

• SIP server hostname or IP address (e.g., sip.provider.com)
• SIP port (typically 5060 for UDP, 5061 for TLS)
• Authentication username and password
• Registration method — IP-based (no registration) or registration-based (username/password)
• Codec support — G.711 (ulaw/alaw) is standard; confirm G.729 if you need bandwidth savings
• Concurrent channel limit — maximum simultaneous calls allowed on your account
• Outbound caller ID numbers — verified DIDs you are authorized to present
• Inbound DIDs — phone numbers routed to your trunk for inbound campaigns

If you are evaluating carriers as part of your vicidial hosting decision, ask providers about per-minute rates, channel limits, and whether they offer dedicated IPs for IP-authenticated trunks. These details affect both vicidial pricing and call quality at scale.

Creating a Carrier Record in VICIdial

VICIdial manages SIP trunks through carrier records in Admin → Carriers. Each carrier record defines how Asterisk connects to an external SIP provider.

1. Navigate to Admin → Carriers → Add a New Carrier
2. Set Carrier ID to a short identifier (e.g., TELNYX, TWILIO, FLOWROUTE)
3. Set Carrier Name to a descriptive label for internal reference
4. Enter the Carrier Description with account details and support contact
5. Set Registration String if your carrier requires SIP registration (format: username:password@host:port)
6. Configure Account Entry for IP-authenticated trunks (host:port without credentials)
7. Set Protocol to SIP
8. Set Active to Y
9. Save the carrier record

For registration-based trunks, Asterisk sends periodic REGISTER messages to keep the session alive. Verify registration status by running sip show registry or pjsip show registrations from the Asterisk CLI on your vicidial server. A successful registration shows State as Registered.

Configuring Servers and Dial Prefixes

After creating the carrier, link it to your VICIdial server and define how outbound calls route through it. Navigate to Admin → Servers and confirm your server record exists with the correct IP address and Asterisk version.

Dial Prefix Setup

Dial prefixes tell VICIdial which carrier to use for outbound calls. In Admin → Carriers → Carrier Detail, configure the dial prefix—a numeric code agents or the auto-dialer prepend to reach that carrier. A common convention uses 9 as the outbound prefix (agents dial 9 + phone number) or routes calls automatically through the campaign's dial prefix setting.

• Dial Prefix — Numeric code associated with this carrier (e.g., 9, 91, 011)
• Dial Plan — Pattern matching for number formatting before sending to the carrier
• Carrier Active — Must be Y for the carrier to accept traffic
• Server IP — The VICIdial server IP that will send calls to this carrier

Campaign-level dial prefix settings override defaults. When configuring your outbound campaign, set the campaign dial prefix to match your carrier configuration so auto-dialed calls route correctly without agent intervention.

Outbound Routes and Number Formatting

SIP carriers expect phone numbers in specific formats—some want 10-digit domestic numbers, others require E.164 format with country code. Misformatted numbers are the leading cause of failed outbound calls after carrier setup.

Common Number Formats

• 10-digit domestic — 5551234567 (most US carriers)
• 11-digit with country code — 15551234567
• E.164 international — +15551234567 (less common in VICIdial, usually stripped to digits)
• With dial prefix — 95551234567 (prefix 9 + 10-digit number)

Configure strip digits and prepend settings in the carrier record to transform VICIdial's internal number format into what your carrier expects. Test with a single outbound call before enabling auto-dialing. If calls connect but audio is one-way, the issue is usually RTP-related rather than number formatting—see our one-way audio troubleshooting guide.

Caller ID Configuration

Outbound caller ID determines what number prospects see when your vicidial dialer calls them. Incorrect or unverified caller ID leads to low answer rates and spam labeling.

Set caller ID at two levels: the carrier trunk (default CID for all calls on that trunk) and the campaign level (overrides trunk default for specific campaigns). Use only numbers your carrier has verified for your account. Rotating through unverified numbers triggers carrier rejections and damages your reputation with callee carriers.

For teams concerned about spam flags on outbound numbers, combine verified caller ID with the practices in our spam labeling prevention guide. Some vicidial hosting providers include CNAM registration and STIR/SHAKEN attestation as part of their managed service.

Inbound DID Routing

If your SIP carrier provides inbound phone numbers (DIDs), configure them in Admin → Inbound → Inbound Groups or DID entries. Inbound DIDs route incoming calls to specific campaigns, IVR menus, or agent groups. Each DID must be configured both at the carrier portal (pointing to your vicidial server IP) and in VICIdial (defining what happens when the call arrives).

Confirm your firewall allows inbound SIP traffic from the carrier's IP ranges. IP-authenticated trunks require the carrier to whitelist your vicidial server IP, and your server must accept traffic from the carrier's SIP gateway IPs.

Failover and Redundancy

Production call centers should never rely on a single SIP carrier. VICIdial supports multiple active carriers with different dial prefixes, enabling manual or scripted failover when a trunk goes down.

Failover Strategies

1. Configure a secondary carrier with a different dial prefix (e.g., primary prefix 9, backup prefix 8)
2. Monitor primary carrier registration status via Asterisk CLI or automated scripts
3. Switch campaign dial prefix to the backup carrier if the primary trunk fails
4. Consider geographic redundancy—use carriers with POPs in different regions
5. Test failover quarterly to confirm the backup trunk is still registered and functional

Some managed vicidial hosting setups include automatic carrier failover as part of their infrastructure. When comparing vicidial pricing across providers, ask whether multi-carrier redundancy is included or requires separate carrier contracts.

Testing Your SIP Trunk

After configuration, run through this validation checklist before routing production traffic through the new carrier.

1. Verify SIP registration shows Registered in Asterisk CLI (sip show registry)
2. Place a manual outbound test call from an agent phone—confirm two-way audio
3. Verify caller ID displays correctly on the receiving phone
4. Place an inbound call to a configured DID—confirm it routes to the correct destination
5. Run 10+ simultaneous test calls to verify channel limits are adequate
6. Check Asterisk logs (/var/log/asterisk/full) for SIP errors during test calls
7. Confirm RTP audio flows in both directions (no one-way audio)
8. Enable a small test campaign and verify auto-dialed calls connect through the carrier

Troubleshooting SIP Issues

Registration Failure

If the carrier shows Unregistered in Asterisk CLI, verify credentials, hostname, and port. Check that your vicidial server IP is whitelisted at the carrier for IP-authenticated trunks. Firewall rules blocking outbound UDP 5060 are a frequent culprit on newly provisioned servers. Restart Asterisk after credential changes: service asterisk restart from the command line.

Calls Fail Immediately (Fast Busy or SIP Error)

Fast busy usually means the carrier rejected the call before it was placed. Check number formatting, verify you have available channels on your account, and confirm the destination number is routable through your carrier. Review the Asterisk CLI output during a failed call for specific SIP response codes—404 (not found) and 403 (forbidden) point to formatting and authorization issues respectively.

One-Way Audio on Carrier Calls

One-way audio after successful SIP setup indicates an RTP or NAT problem. Ensure RTP ports (UDP 10000–20000) are open on your firewall, confirm externip or external_media_address is set correctly in Asterisk if your server is behind NAT, and verify your carrier supports the codec negotiated by Asterisk. Our detailed one-way audio guide covers NAT traversal and RTP debugging.

Calls Dropping Under Load

If calls drop when concurrent volume increases, you may be hitting carrier channel limits or server resource constraints. Check your carrier portal for channel utilization and review server CPU and memory during peak dialing. See why calls drop in VICIdial for a systematic approach to diagnosing drop patterns.

Conclusion

SIP carrier configuration is the bridge between your vicidial server and the public telephone network. Take time to verify registration, test number formatting, configure caller ID correctly, and set up failover before launching production campaigns. A stable SIP trunk combined with proper outbound campaign settings and clean lead imports gives you a dialing operation that agents and supervisors can trust.

For complex multi-carrier setups or persistent SIP issues, engage your vicidial support provider with Asterisk log excerpts and the specific SIP response codes you are seeing. Most carrier problems are solvable within a few hours once the right diagnostic data is available.