星号AMI发起呼叫

Question

我已经使用cisco适配器配置了模拟本地电话，因此可以从SIP电话发出任何出站呼叫。 但是我无法通过AMI实现此功能，AMI通过中继线呼叫出站通道，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UserID，而7000000是样本手机号码。

这是日志：

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

编辑：如何配置我的cisco适配器以重定向传出呼叫，而不是接听并建立网桥？

Answer 1

您已经在ATA上配置了振铃，应答和忙碌识别。

根据您的要求，我可以从星号中看到星号。

如果适配器未通话，则需要检查您的适配器设置。 例如，它可能正在发出声音，为什么您认为这是脉冲。

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