星号 AMI 发起呼叫

Question

我已经使用思科适配器配置了模拟本地电话，因此我可以从SIP电话拨打任何出站电话。但是我无法通过 AMI 实现这一点，AMI 通过中继调用出站通道然后播放提示。
manager.conf ：

[asteriskjava]
secret = asteriskjava  
deny = 0.0.0.0/0.0.0.0  
permit = 127.0.0.1/255.255.255.0  
read = all  
write = all  

extensions.conf ：

[bulk]
exten => 8,1,Playback(thank-you-cooperation)
exten => h,1,Hangup  

source code ：

public class HelloManager
 {
    private ManagerConnection managerConnection;
    public HelloManager() throws IOException
    {
      ManagerConnectionFactory factory = new ManagerConnectionFactory(
            "localhost", "asteriskjava", "asteriskjava");
      this.managerConnection = factory.createManagerConnection();
    }
    public void run() throws IOException, AuthenticationFailedException,
        TimeoutException
    {
      OriginateAction originateAction;
      ManagerResponse originateResponse;
      originateAction = new OriginateAction();
      originateAction.setChannel("SIP/405/7000000");
      originateAction.setContext("bulk");
      originateAction.setExten("8");
      originateAction.setPriority(new Integer(1));
      originateAction.setAsync(true);
      // connect to Asterisk and log in
      managerConnection.login();
      // send the originate action and wait for a maximum of 30 seconds for Asterisk
      // to send a reply
      originateResponse = managerConnection.sendAction(originateAction, 30000);
      // print out whether the originate succeeded or not
      System.out.println("---" + originateResponse.getResponse());
      // and finally log off and disconnect
      managerConnection.logoff();
    }
}  

其中405是用于传出呼叫的 CISCO 适配器的用户 ID，7000000是示例手机号码。

以下是日志：

== Manager 'asteriskjava' logged on from 127.0.0.1
    == Manager 'asteriskjava' logged off from 127.0.0.1
    == Using SIP RTP CoS mark 5
         > Channel SIP/405-0000000c was answered.
      -- Executing [8@bulk:1] Playback("SIP/405-0000000c", "thank-you-cooperation") in new stack
      -- <SIP/405-0000000c> Playing 'thank-you-cooperation.gsm' (language 'en')
      -- Auto fallthrough, channel 'SIP/405-0000000c' status is 'UNKNOWN'
      -- Executing [h@bulk:1] Hangup("SIP/405-0000000c", "") in new stack
    == Spawn extension (bulk, h, 1) exited non-zero on 'SIP/405-0000000c'  

我认为SIP/405正在回答，执行Playback然后挂断，而不是重定向到样本编号。
有什么建议吗？

编辑：如何配置我的思科适配器以重定向传出呼叫，而不是应答和建立网桥？

Answer 1

已在 ATA 上配置振铃、应答和忙识别。

星号按照您的要求工作，据我从您的踪迹中看到。

如果适配器未调用，则请检查您的适配器设置。例如，它可以是用音调调用，为什么你线期望它是脉冲。

也可能是任务的适配器类型不正确。要通过PSTN线路呼叫，您需要FXO适配器，而不是FXS。