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.

Common issues to consider:

• Insufficient Bandwidth: Hosted Voice requires adequate upstream and downstream internet speeds for high-quality calls. When your network's bandwidth is full, or you have network instability, audio quality falls back to the most reliable, lower-bandwidth audio formats so as not to drop the call.
• Network Jitter: Calls will be choppy if packets are delayed or arrive out of sequence due to network congestion. Network jitter is the primary cause of most call quality issues.
• Packet Loss: Network issues may cause data packets to be dropped entirely. This significantly impacts call quality.
• Improper QoS Settings: Failing to prioritize voice traffic can lead to choppy call quality due to packet loss and jitter as other network tasks compete for bandwidth.

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.

Fixes for Choppy Audio

• Enable Quality of Service (QoS) on your router so voice traffic gets priority. This prevents voice packets from being delayed and helps calls sound smoother. Modern routers, modems, and switches allow administrators to implement QoS protocols prioritizing voice traffic.
• Use a wired Ethernet connection rather than WiFi whenever possible. Wireless is more prone to interference, which introduces latency and jitter. Wired connections are more reliable for crystal-clear calls.
• Disable unused network hardware, such as VPNs, guest networks, and high-capacity applications like YouTube, Netflix, and other streaming services, or double NAT configurations that could impact call quality over the network. Reduce internal network congestion by making calls within your office space.
• Place your phones on a Virtual Local Area Network (VLAN) to segment them from other network traffic.

Fixes for Dropped Calls

• Reset Cloud PBX  phones to re-register them with the server. Power cycling fixes registration timeouts that may be causing dropped calls.
• Check that firewalls, VPNs, or routers are not blocking access to critical Cloud PBX ports and protocols. SIP ALG settings may need to be toggled.
• Adjust router settings to allow longer UDP timeouts or switch devices to use TCP.
• Monitor your network quality during calls using built-in phone graphs and logs. Packet loss is a major culprit for call failures.
• Use Power over Ethernet (PoE) switches to hardwire IP phones. This provides consistent connectivity and power in case of WiFi drops or electrical outages.   

Fixes for echo/feedback

Observe if the echo only occurs under certain conditions or directions to narrow down the root cause before troubleshooting.

• Use Proper Cloud PBX Headset/Mic Setups: Use equipment designed for Hosted Voice systems and position microphones away from speakers. If available, enable echo cancellation features.
• Fix Audio Configuration Issues: Misconfigurations commonly cause echo. Correctly calibrate volume levels, gain settings, and other parameters. An experienced Cloud PBX technician can troubleshoot misconfiguration. Configure audio parameters like jitter buffers, packet sizes, and codecs used optimally for your network.
• Enable QoS Settings and VLAN assignments to prioritize voice traffic on your network, minimizing latency, which can exacerbate echo issues.

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.