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SIP trunking in DialNexa lets a workspace connect an existing phone number with username and password authentication or IP-based authentication while still using DialNexa agents, transcripts, recordings, call history, and monitoring. DialNexa SIP trunk modal with IP-based authentication selected and signaling IP allowlist guidance.
SIP trunking gives you control over the telephony path. It also gives you the homework: prove the path works before live traffic.

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Before You Begin

Collect the SIP phone number, termination URI, authentication method, expected caller ID behavior, and one published agent version for testing. For IP-based authentication, ask DialNexa support for the signaling IP used in your region and have the SIP provider allow it before saving the trunk.

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DialNexa SIP Trunking Versus Number Purchase

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Fields In The SIP Trunk Modal

IP-based authentication sends no SIP username or password. The provider authorizes outbound calls by the DialNexa signaling source IP. If provider IPs are not configured for inbound matching, outbound calls can work while inbound INVITEs cannot be mapped to the trunk.
1

Open Phone Numbers

Use the Phone Numbers page to start the SIP linking flow.
2

Choose the authentication method

Use Username & Password when the provider issued SIP credentials. Use IP-based authentication only after the provider has allowlisted the DialNexa signaling IP.
3

Enter the SIP details

Provide the phone number and termination URI. Enter both credentials for username and password authentication. Leave both absent for IP authentication.
4

Assign published versions

The number still needs inbound or outbound published agent versions. SIP decides the route, not the conversation behavior.
5

Place an inbound test

Confirm the correct agent answers, the caller hears audio, and Call History stores the call.
6

Place an outbound test

Confirm caller ID, audio quality, transcript quality, status, duration, and end reason.

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SIP Audio Review

SIP-routed calls use phone-grade audio. The exact quality depends on your carrier and trunk configuration.

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SIP Failures Before Ringing

Both reasons produce a failed call before a useful conversation starts. Correct the route, then place a new test call. Repeated retries without a configuration change are unlikely to help.

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Troubleshooting

Check phone number format, termination URI, authentication method, and whether the linked number is active. Username and password mode requires both credentials. IP mode rejects credentials and requires provider allowlisting.
For IP-authenticated trunks, confirm the provider’s signaling IPs are configured for inbound endpoint matching and that the provider sends traffic to the expected DialNexa route.
Review trunk media settings, recording, and transcript. Compare against a Plivo or web call if you need a baseline.
Phone number assignment controls the agent. Check inbound and outbound published versions.
Test transfer behavior through the exact trunk before production use.

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Verify The SIP Trunk

Place one controlled test call. Confirm that the correct agent answers, both sides can hear clearly, caller ID is expected, and Call History stores the correct direction, route, status, and recording.

Phone Numbers

Manage number status and assignments.

Audio Quality

Review recording and transcript quality.

Call Transfer

Test handoffs through the trunk.

Call History

Audit SIP-routed calls.