Asterisk AMI 发起呼叫

Question

我已经用 cisco 适配器配置了模拟本地电话，所以我可以从 SIP 电话拨打任何呼出电话。但我无法通过 AMI 实现这一点，它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 是用于拨出呼叫的 CISCO 适配器的用户 ID，7000000 是示例手机号码。

这里是日志：

== 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 正在接听，执行Playback 然后挂断，而不是重定向到样本号。
有什么建议？

编辑：如何配置我的 cisco 适配器以重定向拨出电话，而不是接听和建立桥接？

Answer 1

您已在 ATA 上配置响铃、接听和忙碌识别。

就我从您的跟踪中看到的而言，Asterisk 可以按照您的要求工作。

如果适配器没有调用，您已经检查了您的适配器设置。例如，它可以是音调，为什么你的线路期望它是脉冲。

也可能是您任务的不正确适配器类型。通过 PSTN 线路呼出需要 FXO 适配器，而不是 FXS。