SIP Trunking In DialNexa

Connect and test a SIP trunk in DialNexa with a termination URI, credentials, phone number, agent route, and audio review.

SIP trunking in DialNexa lets a workspace connect an existing phone number to a SIP trunk path while still using DialNexa agents, transcripts, recordings, call history, and monitoring.

SIP trunking gives you control over the telephony path. It also gives you the homework: prove the path works before live traffic.

Before You Begin

Collect the SIP phone number, termination URI, authentication requirements, expected caller ID behavior, and one published agent version for testing. Confirm that your SIP provider allows the intended inbound or outbound route.

DialNexa SIP Trunking Versus Number Purchase

Option	Use it when	Operational responsibility
Plivo number purchase	You want to buy and route a number inside DialNexa.	Number availability, compliance, provider readiness, and assignment.
SIP trunking	You already manage telephony and want DialNexa to connect to that route.	Termination URI, credentials, carrier behavior, caller ID, media path, and audio quality.

Fields In The SIP Trunk Modal

Field	Required	Purpose
Phone Number	Yes	The E.164 number to link, such as [phone].
Termination URI	Yes	The SIP destination URI used for routing.
SIP Trunk Username	Optional	Username for trunks that require credentials.
SIP Trunk Password	Optional	Password for trunks that require credentials.
Nickname	Optional	Human-readable label for the number.
SIP rate	Display only	Shows the per-minute SIP trunking rate before saving.

Link And Validate A SIP Number

1. Open Phone Numbers

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

2. Enter the SIP details

   Provide phone number, termination URI, and optional credentials if your trunk requires them.

3. Assign published versions

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

4. Place an inbound test

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

5. Place an outbound test

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

SIP Audio Review

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

Evidence	What to inspect
Recording	Noise, clipping, silence, echo, one-way audio, or bridge issues.
Transcript	Whether Deepgram or Soniox heard the caller correctly.
Call status	Whether the call connected, failed, ended early, or hit voicemail.
Transfer detail	Whether human handoff behaves differently through the trunk.
Audio Cache tab	Whether repeated speech was served quickly or missed cache.

Troubleshooting

Calls fail immediately

Check phone number format, termination URI, credentials, and whether the linked number is active.

Audio is one-way or distorted

Review trunk media settings, recording, and transcript. Compare against a Plivo or web call if you need a baseline.

Wrong agent answers

Phone number assignment controls the agent. Check inbound and outbound published versions.

Transfers behave differently

Test transfer behavior through the exact trunk before production use.

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.

Related Reading

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.