Dialing SIP phone using call file

Asterisk Asterisk Support
1

Hi ,

I have setup asterisk on 1.1.6.2.7 on ubuntu 11.04 .

I have registered a user to asterisk from twinkle sip phone on the same machine where asterisk is running.

in extension.conf i have
[venutest]
exten => 456,1,Dial(SIP/venu@test.com)

In sip.conf I have

[venu]
; Turn off silence suppression in X-Lite (“Transmit Silence”=YES)!
; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
context=venutest ; Where to start in the dialplan when this phone calls
type=friend
regexten=456 ; When they register, create extension 1234
callerid=venu
secret=venu123
host=dynamic ; This device needs to register
nat=no ; X-Lite is behind a NAT router
directmedia=yes ; Typically set to NO if behind NAT
disallow=all
allow=gsm ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw
registertrying=yes ; Send a 100 Trying when the device registers.
username=venu[@test.com]

in call file I have this

Channel: SIP/456
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: venutest
Extension: 456

I am also running this program to capture call details.

package asterisksamples;

import org.asteriskjava.live.AsteriskChannel;
import org.asteriskjava.live.AsteriskQueue;
import org.asteriskjava.live.AsteriskQueueEntry;
import org.asteriskjava.live.AsteriskServer;
import org.asteriskjava.live.AsteriskServerListener;
import org.asteriskjava.live.DefaultAsteriskServer;
import org.asteriskjava.live.ManagerCommunicationException;
import org.asteriskjava.live.MeetMeRoom;
import org.asteriskjava.live.MeetMeUser;
import org.asteriskjava.live.internal.AsteriskAgentImpl;

public class CallDetails {

private AsteriskServer asteriskServer;

void init() {

    asteriskServer = new DefaultAsteriskServer("localhost", "manager", "pa55w0rd");
  asteriskServer.removeAsteriskServerListener(new AsteriskServerListener() {

        public void onNewAsteriskChannel(AsteriskChannel ac) {
          System.out.println("onNewAsteriskChannel ");
          System.out.println("Caller Id " +  ac.getCallerId());
        }

        public void onNewMeetMeUser(MeetMeUser mmu) {
             System.out.println("onNewMeetMeUser ");
        }

        public void onNewAgent(AsteriskAgentImpl aai) {
           System.out.println("onNewAgent ");
        }

        public void onNewQueueEntry(AsteriskQueueEntry aqe) {
             System.out.println("onNewQueueEntry ");
        }
    });
}

public void run() throws ManagerCommunicationException {
    for (AsteriskChannel asteriskChannel : asteriskServer.getChannels()) {
        System.out.println(asteriskChannel);
        System.out.println("Caller Id " +  asteriskChannel.getCallerId());
    }

    for (AsteriskQueue asteriskQueue : asteriskServer.getQueues()) {
        System.out.println(asteriskQueue);
    }

    for (MeetMeRoom meetMeRoom : asteriskServer.getMeetMeRooms()) {
        System.out.println(meetMeRoom);
    }
}

}

Problem when I copy the call file into /var/spool/asterisk/outgoing I dont see call being invoked to sip phone.
I see this in log of the sample I am running.

20 Jan, 2011 7:43:09 AM org.asteriskjava.live.internal.ChannelManager handleNewChannelEvent
INFO: Ignored NewChannelEvent with empty channel name (uniqueId=1295489589.3)

Can anyone help me resolving the issue.

2

The source channel is SIP/venu, not SIP/456. Channel really means device here.

3

thank u david, it works now