Webrtc 使用JSSIP从标头传入PSTN的CallerID

我正在使用JsSIP连接到FreeSwitch，然后连接到PSTN。我想在From头中传递callerID。我的代码设置如下：

var TheCallerIDTest = '+33...number in E164 format';
var TheSipClient = new JsSIP.UA({....}); //works fine

var TheHandlers = {

    'sending': function (e) {

        var TheSipURI = new JsSIP.URI('sip', TheCallerIDTest, 'MyFreeswitchServerUrl', 5060, null, null);
        var TheHeader = new JsSIP.NameAddrHeader(TheSipURI, '', null);

        //displays the correct From header just fine
        console.log(TheHeader);

        //here's where I want to modify the INVITE request
        e.from = TheHeader; 
    },
}

var TheCallOptions = {

    'eventHandlers': TheHandlers,
    'mediaConstraints': { 'audio': true, 'video': false }
};

function TestCall() {

    TheSipClient.start();

    TheSipClient.call("+33...E164 number", TheCallOptions);
}

e.request.headers.From[0] = TheHeader;

查看文档，我希望将

JsSIP.NameAddrHeader

添加到

JsSIP.OutgoingRequest INVITE

请求的“from”头中。控制台输出记录我要添加的正确From头

但是，当我查看

JsSIP:RTCSession emit“sending”[request:InitialOutgoingInviteRequest

控制台日志时，它没有显示我要添加的头，并且服务器上接收到的From头不是我要发送的头

---

我需要在代码中更改什么才能使其正常工作？

您需要这样做：

var TheCallerIDTest = '+33...number in E164 format';
var TheSipClient = new JsSIP.UA({....}); //works fine

var TheHandlers = {

    'sending': function (e) {

        var TheSipURI = new JsSIP.URI('sip', TheCallerIDTest, 'MyFreeswitchServerUrl', 5060, null, null);
        var TheHeader = new JsSIP.NameAddrHeader(TheSipURI, '', null);

        //displays the correct From header just fine
        console.log(TheHeader);

        //here's where I want to modify the INVITE request
        e.from = TheHeader; 
    },
}

var TheCallOptions = {

    'eventHandlers': TheHandlers,
    'mediaConstraints': { 'audio': true, 'video': false }
};

function TestCall() {

    TheSipClient.start();

    TheSipClient.call("+33...E164 number", TheCallOptions);
}

e.request.headers.From[0] = TheHeader;

另外，删除一些不想发送的其他头也可能有意义