JsSIP DTMF Issue with Spy/Whisper/Barge Feature

[Sandaru-Shashinda]

JsSIP DTMF Issue with Spy/Whisper/Barge Feature

I'm attempting to implement FreePBX's spy/whisper/barge functionality in a web application using JsSIP, but having issues with the DTMF functionality.

FreePBX Workflow

As per the FreePBX documentation:

FreePBX Feature code prefix allows spy/whisper/barge on the specified extension.

Usage:

• Dial local extension with 556 prefix to spy
• While spying on active channel use the following DTMF input to toggle modes:
  • DTMF 4 - spy mode
  • DTMF 5 - whisper mode
  • DTMF 6 - barge mode

Current Implementation

I'm currently using JsSIP to connect to our FreePBX server and trying to implement the whisper functionality:

init: async () => {
  if (ua && ua.isConnected()) return;

  JsSIP.debug.disable("JsSIP:*");  

  const session = await getSession();
  if (!session) throw new Error("No active session found. Please log in.");

  const sipExtension = session.user.sip_config.sip_extension;
  const sipSecret = session.user.sip_config.sip_secret;

  if (!sipExtension || !sipSecret)
    throw new Error("SIP credentials not found in session.");

  const socket = new JsSIP.WebSocketInterface("wss://domain/ws");
  const configuration = {
    sockets: [socket],
    uri: `sip:${sipExtension}@[domain]`,
    password: sipSecret,
    display_name: "Client",
  };

  ua = new JsSIP.UA(configuration);

  // Various event handlers...
  ua.on("registered", () => {
    status = "Connected to PBX";
    // Successfully registered
  });

  ua.on("newRTCSession", (data) => {
    // Session handling...
  });

  ua.start();
},

whisperCall: async (sipConfig) => {
  console.log("Whispering to:", sipConfig);

  if (!ua)
    throw new Error("SIP user agent is not initialized. Call init first.");

  if (currentSession)
    throw new Error(
      "Another call is in progress. End the current call first."
    );

  const targetUri = `sip:${sipConfig.sip_extension}@${SIP_DOMAIN}`;

  // Store the session from the call
  currentSession = ua.call(targetUri);

  // Add event listener for when the call is connected
  currentSession.on("confirmed", () => {
    // Only send DTMF after the call is established
    currentSession.sendDTMF(5, { transportType: "RFC2833" });
    console.log("DTMF tone sent");
  });

  if (!currentSession) throw new Error("Failed to initiate whisper.");

  return currentSession;
}

Problem

1. When I establish the call using JsSIP, I'm not sure if I need to prefix the extension with "556" as would be done with a regular phone, or if I need to handle that in the SIP URI structure.

2. When I attempt to send DTMF tone "5" to enter whisper mode after the call is established, it doesn't appear to be recognized by the FreePBX server.

3. When my agent is in a call with a client as an admin I want to whisper to my agent

Questions

1. What is the correct way to implement the FreePBX spy/whisper/barge feature using JsSIP?

2. Should I be dialing with the prefix in the SIP URI (e.g., sip:556${extension}@${domain}) or should I dial the extension normally and then use DTMF?

3. Are there specific JsSIP settings or configurations needed for DTMF to work correctly with FreePBX?

Environment

• JsSIP version: 3.10.1

Any guidance on the correct implementation would be greatly appreciated.