SIP Trunk Configuration - Documentation | AnrizVoIP Contact Center

SIP trunks connect your platform to the telephone network, enabling inbound and outbound calling.

Supported Providers

The platform works with any standard SIP trunk provider:

Twilio

Vonage (Nexmo)

Telnyx

Bandwidth

VoIP.ms

Flowroute

And many others

Navigate to Telephony > SIP Trunks

Click "Add Trunk"

Enter the trunk details from your provider:

Trunk Name (for identification)

SIP Server / Proxy address

Port (usually 5060 for UDP, 5061 for TLS)

Authentication (username/password or IP-based)

Codec preferences (G.711, G.729, Opus)

Click "Save and Test"

DIDs (Direct Inward Dialing) are phone numbers assigned to your platform:

Navigate to Telephony > DIDs

Click "Add DID"

Enter the phone number from your trunk provider

Assign it to a tenant, queue, or IVR

Configure caller ID for outbound calls

After configuration:

Make a test outbound call to verify connectivity

Call your DID number to test inbound routing

Check audio quality (no echo, clear voice, no drops)

Verify caller ID displays correctly

Common issues:

No audio: Check firewall rules for RTP ports (typically 10000-20000)

One-way audio: NAT configuration may need adjustment

Registration failures: Verify credentials and server address

Call drops: Check network stability and bandwidth