Logging is used to check real-time call attempts, SIP traces, routing status, and to simulate a call. As soon as a call hits the ConnexCS system, it will display in the Logging area. This can be accessed via Management > Logging.
To view calls that are having issues registering, click
Register Logging, then click on a specific Call ID to view Call Details and SIP Trace.
Call ID Details¶
Click on a specific Call ID to view details and run call tools.
- Call Details: The initial screen shows current details which include Routing Status, Authentication, Induced PDD, RTP, Routing Engine ID, DTMF, as well as several other pieces of information. There are also graphs that show stats such as Jitter and Packet Loss for the call.
- Raw Data: Underlying data that populates for the call.
- SIP Trace: Visual representation of SIP communications, see details below under SIP Traces.
- Simulate: See details below under Simulating Calls.
- Refresh: For Live calls, use the
Refreshbutton to reload the logs to show the most recent changes. This is necessary as some of the data must be processed through CDR before it will be displayed.
SIP Tracing is a tool used for phone systems that use the SIP (Session Initiation Protocol) for interactions like trunks and endpoints. Traces give detailed information about calls and call attempts which helps to identify any problem areas for debugging and troubleshooting.
To view the SIP Trace of a call:
- Click a Call ID to view its SIP traces.
SIP tracesbutton to view the SIP trace.
Toggle between Relative Time and Absolute Time for specific time of day
- Options to download as Text or an Image
Known issues with SIP Traces
- Missing SIP data: SIP traces are not always guaranteed. SIP packets are carried by UDP, which may cause the traces to be lossy at times. This is to be expected due to the nature of the architecture.
- Missed call attempts: If using SIP authentication, because there are 2 requests it is possible that they hit our database out of order. This may cause the logging page to only display the first call attempt.
- These are considered reporting calls, and do not impact the calls directly. They are both rare, typically observed in less than 1 in every 50,000 calls
Simulating calls allows providers to identify areas of concern, or just to verify functionality, by testing in different setups and operational configurations.
To simulate calls:
Simulateeither from the Logging screen or from within a specific Call ID. A form will appear:
Enter the dialed number and CLI/ANI number
- Select a switch IP from the available options.
- Enter the IP address.
- Optionally, add the SIP extension user
- Select the routing engine zone
The simulation call result will appear in logging. The Call ID will be prepended with a
SIM tag. Click the Call ID to view the call's routing status.
Testing a fixed issue
After you have fixed a routing issue with a specific call, you can go into the Call ID and run the Simulate tool to make sure any routing issues are resolved and now pass.
Searching the Logs¶
You can search for calls by phone number, Call ID, or IP address, by entering one of these attributes into the text box at the top-right of the Logging page and clicking the