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.