When using SIPSorcery with .NET 6 to make SIP calls via VoIP.ms, the custom Caller ID is not being passed correctly, despite being set in the code.

We have the setting configured in VoIP.ms as:
“I use a system capable of passing its own CallerID”

using ConsoleApp.Sipsorcery;
using Microsoft.Extensions.Logging;
using Microsoft.Extensions.Logging.Abstractions;
using Serilog;
using Serilog.Extensions.Logging;
using SIPSorcery.Media;
using SIPSorcery.SIP;
using SIPSorcery.SIP.App;

public class Program
{
    private static Microsoft.Extensions.Logging.ILogger _logger = NullLogger.Instance;

    const string destination = "DESTINATION_NUMBER";
    const string username = "USERNAME";
    const string password = @"PASSWORD";
    const string domain = "xxxxx.voip.ms";

    // Internal variable for Caller ID
    private static string callerId = "OWN_CALLER_ID"; 

    public static async Task Main(string[] args)
    {
        AddConsoleLogger();

        _logger.LogInformation("SIP Get Started");

        var userAgent = new SIPUserAgent();
        var voipMediaSession = new VoIPMediaSession();
        voipMediaSession.AcceptRtpFromAny = false;
        voipMediaSession.AudioExtrasSource.SetSource(AudioSourcesEnum.Silence);

        SipDialerAccount sipDialerAccount = new SipDialerAccount();
        int maxRetries = 5;
        int retryCount = 0;
        bool callResult = false;

        while (retryCount < maxRetries)
        {
            try
            {
                SIPCallDescriptor callDescriptor = new SIPCallDescriptor(
                    username: username,
                    password: password,
                    uri: $"sip:{destination}@{domain}",
                    from: $"sip:{callerId}@{domain}",
                    to: $"sip:{destination}@{domain}",
                    routeSet: null,
                    customHeaders: null,
                    authUsername: null,
                    callDirection: SIPCallDirection.Out,
                    contentType: null,
                    content: null,
                    mangleIPAddress: null);

                callResult = await userAgent.Call(callDescriptor, voipMediaSession, 0);

                if (callResult)
                {
                    _logger.LogInformation("Call succeeded.");
                    break;
                }
                else
                {
                    _logger.LogWarning($"Call failed. Retrying... (Attempt {retryCount + 1} of {maxRetries})");
                    retryCount++;
                    await Task.Delay(10000); // Delay between retries
                }

            }
            catch (Exception ex)
            {
                _logger.LogError(ex, "Error making call.");
                break;
            }
        }

        if (!callResult)
        {
            _logger.LogError("All call attempts failed.");
        }
    }

    private static void AddConsoleLogger()
    {
        var logger = new LoggerConfiguration()
            .Enrich.FromLogContext()
            .MinimumLevel.Is(Serilog.Events.LogEventLevel.Debug)
            .WriteTo.Console()
            .CreateLogger();
        var factory = new SerilogLoggerFactory(logger);
        SIPSorcery.LogFactory.Set(factory);
        _logger = factory.CreateLogger<Program>();
    }
}

This is the Serilog output:

[22:51:12 INF] SIP Get Started [22:51:12 DBG] CreateRtpSocket attempting to create and bind RTP socket(s) on [::]:0. [22:51:12 DBG] CreateBoundSocket attempting to create and bind socket(s) on [::]:0 using protocol Udp. [22:51:12 DBG] CreateBoundSocket even port required, closing socket on [::]:49667 and retrying on 49668. [22:51:12 DBG] CreateBoundSocket attempting to create and bind socket(s) on [::]:49669 using protocol Udp. [22:51:12 DBG] CreateBoundSocket successfully bound on [::]:49669, dual mode True. [22:51:12 DBG] Successfully bound RTP socket [::]:49668 (dual mode True) and control socket [::]:49669 (dual mode True). [22:51:12 DBG] RTPChannel for [::]:49668 started. [22:51:12 DBG] SIPClientUserAgent attempting to resolve atlanta2.voip.ms. [22:51:13 DBG] SIPClientUserAgent resolved atlanta2.voip.ms to udp:MY_IP:5060 in 1074.87ms. [22:51:13 DBG] UAC commencing call to sip:[email protected]:5060. [22:51:14 DBG] CreateBoundSocket attempting to create and bind socket(s) on 0.0.0.0:0 using protocol Udp. [22:51:14 DBG] CreateBoundSocket successfully bound on 0.0.0.0:57590. [22:51:14 INF] SIP UDP Channel created for udp:0.0.0.0:57590. [22:51:14 DBG] Information response 100 Trying for sip:[email protected]. [22:51:14 INF] Call attempt to sip:[email protected] received a trying response INVITE 100 Trying. [22:51:14 DBG] Response 603 Declined for sip:[email protected]. [22:51:14 WRN] Call attempt was answered with failure response INVITE 603 Declined. [22:51:14 WRN] Call failed. Retrying… (Attempt 1 of 5)

Do I missing something for the callerId ???