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BYO SIP (Bring Your Own SIP) lets you connect a SIP trunk from your existing carrier to DialNexa. Use this when you have an established carrier relationship, negotiated per-minute rates, or geographic coverage that DialNexa’s default providers do not offer. DialNexa handles Voice AI processing while your carrier handles PSTN connectivity.

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When to Use BYO SIP

ScenarioUse BYO SIP
You have a carrier contract with committed minutes or ratesYes
Your region is not served by PlivoYes
Your organization requires a specific carrier for security or complianceYes
You want to consolidate calling through an existing carrierYes
You are migrating from another platform and keeping your carrierYes
You are starting fresh with no carrier preferenceNo (use Plivo by default)

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Prerequisites

  • An active SIP trunk with your carrier, configured for outbound termination and optionally inbound origination.
  • The SIP termination URI provided by your carrier (e.g., sip.yourcarrier.com or a full URI like sip:[email protected]).
  • The phone number(s) you want to use, already provisioned with your carrier.
  • Optional: SIP authentication credentials (username and password) if your carrier requires registration or digest authentication.

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Connect a SIP Trunk

1

Open Telephony Settings

Navigate to Settings > Telephony > SIP Trunks. Click Add SIP Trunk.
2

Enter trunk details

Fill in the following fields:
FieldRequiredDescription
Phone NumberYesThe phone number in E.164 format (e.g., +14155550100). This is the number associated with this trunk.
Termination URIYesThe SIP URI your carrier provided for outbound call termination. Example: sip.carrier.com or sip:[email protected].
UsernameNoSIP authentication username, if your carrier requires it.
PasswordNoSIP authentication password. Stored encrypted.
NicknameYesA label for this trunk used in the DialNexa UI. Example: Carrier XYZ - US East.
3

Review estimated per-minute rate

DialNexa displays an estimated per-minute SIP rate based on the destination region of the termination URI, if available. This is informational. Your actual rate is governed by your carrier contract, not DialNexa.
4

Save and test the connection

Click Save. The trunk is added and a connection test is initiated. DialNexa sends a SIP OPTIONS ping to the termination URI to confirm connectivity. The result appears within a few seconds:
  • Reachable: The carrier responded. The trunk is ready.
  • Unreachable: The OPTIONS ping failed. Check the termination URI and confirm your carrier’s firewall allows SIP from DialNexa’s IP ranges.

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DialNexa IP Ranges for SIP

If your carrier has IP allowlisting on the SIP termination endpoint, add the following DialNexa SIP source IP ranges: Contact DialNexa support for the current list of SIP media and signaling IP ranges for your region. The ranges may change; subscribe to the DialNexa status page for network change notifications.
IP ranges vary by DialNexa region (US, EU, India). Ensure you allowlist the ranges for the region where your DialNexa workspace is hosted.

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Assign the SIP Trunk to a Number

After adding the SIP trunk, the phone number appears in Phone Numbers with BYO SIP as the Telephony Provider. Assign an inbound agent version and an outbound agent version as you would for any other number. See Receive Inbound Calls and Make Outbound Calls.

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Inbound Calls on BYO SIP

For inbound calls on a BYO SIP trunk, your carrier must forward calls to DialNexa’s SIP ingress URI. Configure your carrier’s inbound routing to send SIP INVITEs to: 
sip:[email protected]
Include the dialed number in the To header in E.164 format so DialNexa can match it to the correct phone number and agent version. Contact DialNexa support for the exact inbound SIP URI for your workspace region.

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Per-Minute SIP Rates

DialNexa displays an estimated per-minute rate for your SIP trunk in the trunk detail view. This estimate is based on publicly available carrier rate tables and is not guaranteed. Your actual cost is the rate in your carrier contract. Your total per-call cost for a BYO SIP call: 
Total cost = DialNexa per-minute Voice AI rate + Your carrier's per-minute rate
DialNexa does not invoice or collect your carrier’s portion. Carrier costs are billed directly by your carrier.

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Test the SIP Connection

After adding a SIP trunk:
  1. Open Settings > Telephony > SIP Trunks and locate the trunk.
  2. Click Test Connection to send a fresh SIP OPTIONS ping.
  3. Place a test outbound call using the trunk number as the from_number via the API or dashboard.
  4. Verify the call connects and audio flows in both directions.
Use the dashboard test call feature to place an outbound call to your own mobile phone. This is the fastest way to confirm audio quality end-to-end without setting up additional infrastructure.

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Troubleshooting

Verify the termination URI is correct. Check that your carrier’s SIP endpoint allows traffic from DialNexa’s IP ranges. Some carriers require you to pre-register the trunk or add DialNexa as an authorized peer on their platform.
Audio (RTP) and SIP signaling travel on different paths. If signaling works but audio does not, your carrier may have media IP allowlisting separate from SIP. Confirm the carrier allows RTP from DialNexa’s media IP ranges.
Confirm you have configured your carrier’s inbound routing to forward SIP INVITEs to DialNexa’s inbound SIP URI. The To header must contain the dialed number in E.164 format.
If your carrier requires digest authentication, ensure the username and password in DialNexa match exactly (case-sensitive) what your carrier expects. Some carriers also require the username to include the domain portion (e.g., [email protected]).