Telephony

Telephony configuration connects Dialer.CX to the public telephone network. This section covers SIP trunk management and phone extension provisioning — the foundation of all inbound and outbound calling.

Carriers / Trunks

SIP trunks connect Dialer.CX to the telephone network for outbound and inbound calls. Each carrier represents a connection to a VoIP provider or PSTN gateway.

Managing Carriers

Navigate to Admin > Carriers to manage your SIP trunk inventory:

• Add new carriers with SIP connection details (hostname, port, protocol)
• Configure credentials — username/password or IP-based authentication
• Set codec preferences — audio quality settings (G.711, G.729, etc.)
• Define max channels — concurrent call limits to prevent trunk overload
• Enable/disable carriers instantly without deleting their configuration

Note

When you save carrier changes, Dialer.CX writes the configuration to the database and triggers a PBX config rebuild. The new trunk settings are active within seconds — no manual server restart required.

Carrier Groups

Group carriers for load balancing and failover:

• Primary and backup carriers — automatic failover when the primary is unreachable
• Round-robin distribution — spread calls across multiple trunks evenly
• Weighted routing — send more traffic to higher-capacity or lower-cost carriers
• Automatic failover when a carrier is unavailable or returns SIP errors

Phone Extensions

Manage SIP phone registrations for agents at Admin > Phones:

• Create extensions with unique credentials (extension number, SIP password, caller ID)
• Browser phone configuration — extensions optimized for the built-in WebRTC softphone
• Desk phone support — extensions for physical SIP phones (Polycom, Yealink, Grandstream, etc.)
• Extension monitoring — real-time view of which phones are currently registered and active

Tip

For agents using the browser-based softphone, extensions are automatically provisioned when you set their phone mode to “Browser” during user creation. You can also manually create and assign WebRTC extensions here for fine-grained control.

Best Practices

Tip

• Monitor carrier health regularly — check answer rates, ASR (Average Seizure Ratio), and call quality per carrier
• Set appropriate channel limits to prevent overloading trunks and degrading audio quality
• Use carrier groups for redundancy — never rely on a single trunk for production traffic
• Test new carriers with a small number of calls before routing full campaign volume through them
• Keep codec settings consistent between your carriers and the PBX to avoid unnecessary transcoding