CallerID not working using SIPSorcery .net 6 with Voip.ms

[leandrotyl-convertros]

Hi All!

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 setup in Voip.MS the setting ---> "I use a system capable of passing its own CallerID"

And the code that I'm using is this:

`
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 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:NUMBER_NUMBER@atlanta2.voip.ms: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:NUMBER_NUMBER@atlanta2.voip.ms.
[22:51:14 INF] Call attempt to sip:NUMBER_NUMBER@atlanta2.voip.ms received a trying response INVITE 100 Trying.
[22:51:14 DBG] Response 603 Declined for sip:NUMBER_NUMBER@atlanta2.voip.ms.
[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)

[sipsorcery]

You'll need to dig into the voip.ms docs (or ask ChatGPT) what SIP header they use to set caller ID. It is normally the From header but not always.

Other than that your example From header looks a bit weird, normally it would be:

from: $"\"{callerId}\" <sip:{username}@{domain}>",

[leandrotyl-convertros]

Thanks @sipsorcery for you quick response. I've tried with your changes that you have mentioned but NO luck.

When you said "...what SIP header they use to set caller ID", Can I use customHeaders prop to pass it in that way?

Also is there a way to get more detailed logs (from .net and PBX) from the SIPSorcery library you have created?

Many thanks!

:)

[leandrotyl-convertros]

I made some progress so far with the CLI PJUA with VOIP.ms. The thing is that voip.ms allows to override CallerID when you setup their configuration by: Static IP Authentication (SIP only) (Advanced Users)

( ) User/Password Authentication (Recommended)
(x) Static IP Authentication (SIP only) (Advanced Users)

Here is the command I'm using in PJSUA:

pjsua --id="sip:MY_CALLERID@atlanta1.voip.ms" --registrar="sip:atlanta1.voip.ms" --realm="*" --local-port=5060 sip:DESTINATION_NUMBER@atlanta1.voip.ms

How should I adapt this into Sipsorcery approach?

Best,

:)

[leandrotyl-convertros]

Finally I figure out!!

Voip.ms has a rule that there is no documentation at all, but if you need to override caller_id you need to SETUP voip.ms like this:
(x) Static IP Authentication (SIP only) (Advanced Users)
(x) "I use a system capable of passing its own CallerID"

And in the code replace username with your callerId

bool callResult = await userAgent.Call(DESTINATION, CALLER_ID, null, voipMediaSession);