Overview

If you are experiencing frequent call drops, please refer to the article below to gather the needed information and open a support ticket.

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Prerequisites

• APIM admin role
• Screenconnect access
• PSX access
• Hepic access

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Solution

To start with troubleshooting, we will need call samples. Open a ticket with Support, making sure to describe the issue and any symptoms you observe, and add at least two call samples in the following format:

Call sample 1:

• Calling From Number:
• Calling To Number:
• Date of call: dd/mm/yy
• Time of call: xx:xx am/pm (time zone)
• Result: ex: dropped call, garbled audio, etc.
• Trouble observed: Inbound or Outbound

Call sample 2:

• Calling From Number:
• Calling To Number:
• Date of call: dd/mm/yy
• Time of call: xx:xx am/pm (time zone)
• Result: ex: dropped call, garbled audio, etc.
• Trouble observed: Inbound or Outbound

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The impact of this type of case is usually high as it directly affects the customer's business. Once you have the details and call samples from the customer, proceed with the steps below.

1. Look up the numbers that are not working in the Apimarket. We do this to rule out problems with the portal, as sometimes the number may not have a package assigned to it, which would explain why the number is not working correctly. In the screenshot below, the TN does not have a package assigned:
   
   

   Note: In some other cases, the package can be as unassigned, yet if we can find the TN on PSX (see step 2), it should be fine.

2. Next, we should look up the numbers on the PSX to check if there is a route for them. If the numbers have a package assigned on APIM we should see the route on PSX. If that's the case, proceed to step 3.

   In some cases, even with no package assigned on APIM the route exists in PSX. This type of cases are STaaS, and the number is supposed to work fine for these too, so we can proceed to step 4 as well.

   But, if there is no package assigned on APIM and the number does not have a route on PSX, we should assign a package or create the route on PSX depending on the type of case (DR or STaaS).

3. Next, we should look up for the calls in Hepic from one of the Screenconnect VMs in the corresponding Data Center. Login to Hepic and search based on the customer samples.
   
4. Once the calls are found, check the SIP call flow to find out any problems (refer to SIP Call Flow Explained and SIP Codes). Proceed based on what you discover.

Known Cases

The issue on the customer side

When you see that call has a duration (there are at least several seconds between the last ACK from SBC to ATT and the 200 OK), and the BYE (SIP message) is coming from the customer (IP belonging to the customer) - it means the call had duration and likely the end-user on the customer side just ended the call.

Share the findings with the customer to investigate. For example, in one case the customer discovered that "agent was forwarding this call via warm transfer to the same number that the call originally called in from"

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Based on the troubleshooting the Support team will either take action to fix the issue on our side, or reach back with findings and any next steps required from you.

Verification

Once the root cause is fixed, try to replicate the issue to confirm that it is no longer there.