This article outlines how to troubleshoot outbound or inbound call completion issues such as hearing:“The number you have dialled is not in service.”

• Call drops immediately.
• One-way audio or fast busy signal.

These problems often stem from routing issues, number formatting errors, or carrier-level restrictions.

Step-by-Step Troubleshooting

 1. Confirm the Number Format

Ensure the dialled number is in E.164 or 10-digit format, depending on your system's dialling rules.

Example: +14035551234 or 4035551234

 If using Speed Dials or custom buttons, verify that those are not prepending incorrect digits.

Check the Calls

Use the Call Flow and Gradle to Grave functionality in the portal. Look up the affected call.

The Cradle to Grave screen opens.

SIP Flow and Link: In the portal, navigate to Call History and filter the results to find the calls with quality issues. Select the "SIP Flow" icon to view the SIP flow detail. 

The SIP Flow screen opens. Click the "Share" button to copy the link to your clipboard. If you escalate the issue, retain this link to share with the support team.

Review the call path and SIP response codes (e.g., 404, 503, 488).

Determine which leg of the call failed:

• User to SBC?
• SBC to Carrier?
• Common SIP Errors:
  • 404 Not Found: Number does not exist in a carrier or is in the wrong number format.
  • 503 Service Unavailable: Carrier or route not available.
  • 488 Not Acceptable Here: Media negotiation or codec issue.

 3. Inbound Call to a Number That Fails

• Confirm the DID is assigned to the correct user or call flow (e.g., Auto Attendant, Ring Group).
• Verify the porting status. Has the number been fully ported?

4. Outbound Call Fails

• Test the same number from a different extension.
• Try calling the number from a mobile phone to confirm it is in service.
• Check if calls to similar prefixes (e.g., same area code) are also failing.
• Confirm that the Outbound Dial Plan includes the route to the number's destination.

5. Carrier-Specific Blocks or Routing Issues

Some carriers may block high-cost, international, or high-risk routes.

 6. Media/Audio Issues vs. Signalling

If the call connects but no audio is heard:

• Ensure RTP ports are open (usually UDP 10,000–20,000).
• Check for NAT issues or double NAT.
• Confirm that codecs are compatible on both ends.

7. Confirm Number Status

Use online tools (e.g., FreeCarrierLookup) or porting platforms to verify:

• The number is active.
• Correct carrier of record.
• Whether it's recently ported.

Escalation Criteria

Escalate to support if:

• The issue affects multiple users or numbers consistently.
• Call Trace shows the call leaving your system correctly, but it still fails.
• You suspect that the call is to a high-risk destination, and you are willing to cover the costs of calls to the destination to turn the location on.
• The SIP code returned is ambiguous or intermittent.

Include:

• Dialled number and format.
• Time/date of failed call.
• Call Trace screenshots or the SIP Flow link.
• Affected users and domains.

Summary

The Cradle to Grave screen opens.

SIP Flow and Link: In the portal, navigate to Call History and filter the results to find the calls with quality issues. Select the "SIP Flow" icon to view the SIP flow detail. 

The SIP Flow screen opens. Click the "Share" button to copy the link to your clipboard. If you escalate the issue, retain this link to share with the support team.

Look for a BYE or CANCEL and who sent it. If the BYE comes from the user side, it's likely a device, network, or NAT issue. Investigate upstream routing or the SIP trunk if it comes from the carrier.

Identify SIP error codes (e.g., 408 Request Timeout, 481 Call/Transaction Does Not Exist).

 3. Check Device & Network Environment

• Reboot the affected phone or softphone.
• Verify that SIP registration is stable.
• Ensure no double NAT exists in the network path.
• Check for SIP ALG enabled on the router/firewall (should be disabled).
• Review session timeout settings (e.g., UDP timeout < 60s can cause drops) or use TCP.
• Ping and perform traceroutes to SIP/RTP server addresses.
• Run a packet capture (Wireshark) if available. Packet loss is a major culprit for call failures.

 4. Review Device and Firmware Configuration

• Ensure phone firmware is up to date.
• Check SIP session timers (Session-Expires) and keep-alive intervals.
• For ATAs or SIP endpoints, confirm NAT keep-alive is enabled.
• Validate SIP proxy and outbound proxy settings 

Recommendations

Dropped calls are often linked to signalling path instability, NAT/firewall configurations, or SIP session management. Combining call trace analysis with local network and endpoint inspection allows most dropped call issues to be quickly diagnosed and resolved.

Escalation

If the issue consistently occurs on calls to specific numbers or carriers:

• If you have one, share the Call Trace to extract Call-ID and SIP routing path.
• Record the extensions or numbers affected.
• Record the timestamp and time zone.
• Record symptoms and duration.
• Call Trace link.
• Record any relevant IPs or device models.
• Identify any SIP 5xx or 4xx responses on their leg of the call.