星号AMI发起呼叫

Question

I have configured the analog local phone with cisco adapter, so I can make any outbound call from SIP phone. But I can't achieve this by AMI which calls to outbound channel through trunk then plays prompt.
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();
    }
}  

Where 405 is the UserID of CISCO adapter for outgoing calls, 7000000 is a sample cell phone number.

Here is the logs:

== 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'  

I think SIP/405 is answering, executing Playback then hangs up, not redirecting to sample number.
Any suggestions?

EDIT: How can I configure my cisco adapter in order to redirect outgoing calls, not to answer and make the bridge?

Answer 1

You have configure ring, answer and busy recognition on your ATA.

Asterisk work as you requested as far as i can see from your trace.

If adapter not calling, you have check with your adapater settings. For example it can be calling in tone, why you line expect it is pulse.

Also can be incorrect adapter type for your task. For calling out via PSTN line you need FXO adapter,not FXS.