Config questions

Asterisk Asterisk Support
1

Newbie question:

What asterisk configuration needs to take place in order to originate a call from a Java application.
My SIP trunk is inphonex.com

What I know so far is that the following files to be properly configured.

asterisk.conf
manager.conf
sip.conf.

Is there any other files where a configuration setting is necessary.

Thanks in advance.

2

Extensions.conf, very important, where all the Asterisk’ extensions are defined, including the extensions to call out through a sip trunk.

Cheers.

Marco Bruni

3

Thanks for the reply.
I’m having a difficult time understanding the extensions config part of this.
All I want to do is dial a mobile number from a Java application. There isn’t any extensions involved with doing this.
The entire dial string looks like this

 
               OriginateAction originateAction = new OriginateAction();
	
               //: Where xxxxxxxxxx is the mobile phone number to dial
	       originateAction.setChannel("SIP/xxxxxxxxxx@sip.inphonex.com");
	        originateAction.setCallerId("xxxxxxxxxx");
	        originateAction.setPriority(new Integer(1));
	        originateAction.setTimeout(new Long(30000));
              // send the originate action and wait for a maximum of 30 seconds for Asterisk
	        // to send a reply
	       ManagerResponse originateResponse = managerConnection.sendAction(originateAction, 30000);

	        // print out whether the originate succeeded or not
	        log.info(originateResponse.getMessage());

When this java code is executed.

I’m receiving the following output on the Asterisk CLI console.
== Using SIP RTP CoS mark 5
– Got SIP response 484 “Address Incomplete” back from 208.239.76.169

I’m having a hard time connecting the dots between the Java code above and the
Asterisk config files.

4

Originate need a context, an extension and a priority, you put only the priority, this three parameters tell to Asterisk where to send the call (what actions to do) when the called channel (in your case see SIP/xxxxxxxxxx@sip.inphonex.com) answers, see voip-info.org/wiki/view/Aste … +Originate .

Cheers.

Marco Bruni

5

i have a similar issue. How can i figure out what the Transport/Trunk/Number should be? I have an asterisk box with SIP phones connected to it. There is a SIP trunk to a mitel pbx that provides connection to the PSTN. I can call from asterisk to mitel, from mitel to asterisk, and from asterisk to pstn, so THOSE pieces are working.

when calling from asterisk (1091) to mitel (4035), the cli log shows device SIP/Mitel/4035. when manager originate is attempted, doesn’t like that string (SIP/Mitel/4035). i CAN use that string in a call file, i can also use that string for cli originate.

It is also possible to use “local/4035@mitel-context”, but i think there are issues reported elsewhere with using local (for example, the call seems to be “answered” when passed to the mitel, not when the user at extension 4035 picks up).

What SHOULD the transport/trunk/number string be, since it doesn’t appear to be what appears in cli log.