Hello all,
We have a problem in our CTI-System using AES 6.2 and so our distributor gave us the tip with your developer forum. We use an Avaya VoIP system with an integration of 3rd party CTI service in our lotus notes telephone book. From our distributor we got the Telsvc SDK 6.2.0.69 which is a part of WebService I think. Now we had implemented the functionality to lotus notes and it works beautiful. But we have the following problem.

If a caller calls a callee and the phone of the callee is busy because of an active call, the CTI system abort the call and nothing happens. No message, no signaling on the phone, nothing. Question: Is it possible getting a busy signal on the line when the callee is busy?

Please give us a tip to realize this, because our distributor can't tell us a solution for this problem.

I would start with looking at the TSAPI logs on AE Services. Have a look at the FAQ called "What is the procedure for enabling and accessing the AE Services logs for TSAPI (trace, tracing, g3trace, csta_trace)?"

There is also one called "What is the procedure for enabling and accessing the AE Services logs for the Telephony Service Web Service (trace, tracing)?"

See if these logs will tell you anything. Sounds like the call is getting disconnected before hearing any busy tone feedback, but you will have to try to confirm that with the logs.

If you need us to dig deeper and review your logs please open a ticket with Devconnect and we can supply you with some information to procure some support hours so that we can further assist you.

Also, I was able to set up a test with the TelSvcGuiClient sampleapp that comes with the SDK. I made a call from A -> B and put the call on hold. I made a second call from A -> B and put the call on hold. I then launched the sample application and placed a call from A -> B and I heard a busy tone. Have you tried the sample application that comes will the Telephony Web Service SDK?

When you make the call from one station to the other without the telephony web service, I assume in that case you are hearing dial tone? Then with the same extensions using the telephony web service you don't hear dial tone?

In order to describe our environment in detail:

We use the procedure makeCall of the class TelSvcWrapper included in the TelSvcGuiClient sampleapp. The method makeCall works fine with outbound calls. In case of inbound calls the makeCall throws the InvalidPartyException but we would expect that the CTI System should handle it to the calling telphone by providing a busy dial tone or a redirection to a call center or something like that.

Afterwards a code snippet of the TelScvWrapper class:

/******************************************************************************
// TelSvcWrapper
//******************************************************************************
/**
* Simple wrapper class for the Telephony Service that handles interfacing to
* the AXIS generated classes: setting HTTP and SOAP headers, etc.
*
* @author AVAYA, Inc.
*/

@SuppressWarnings("unused")
public class TelSvcWrapper {
private static TelSvcWrapper telSvcTest;

private TelephonyService port = null;
private TelephonyServiceService service;
private TelephonyServiceSoapBindingStub telSvcSoap = null;

private static final String serviceName =
"http://xml.avaya.com/ws/TelephonyService/2005/04/04";

private static final String sessionNameSpace =
"http://xml.avaya.com/ws/session";
private static final String sessionKey = "sessionID";

It makes no sense to use the "makeCall" service in the context of an inbound call. MakeCall originates a call to a destination number from a specified station on the PBX that is associated with the switch connection link identified by the application.

To me a inbound call is a call that is originated by some other station that is alerting on the station of interest (the target of the CTI operation the application originates).

So what exactly do you mean by inbound? Are you talking about a INTERNAL station to station call within the PBX? If so what is the state of the originating device (can you go off hook and place a call)? What is the state of the destination station (can you go off hook and place a call)?

Sorry about the wrong use of "inbound call", we mean internal call where we have the problem with the InvalidPartyException thrown by method execution of makeCall. We initate the makeCall to a target destination that is busy. We would prefer to get a busy dial tone instead of InvalidPartyException.

Our sample client uses a version from 2005/04/04 as outlined by attribute serviceName within the class TelSvcWrapper.

When you dial the busy station manually from the phone you are using with Telephony Web Services does it give you a busy signal?

On manual dialing from station A to busy B there is a busy tone but not if we dial via CTI over the web service - and that's the problem.

Please login to your AE Services Management Console. Go to Utilities -> Diagnostics -> AE Service -> TSAPI test and run the test from there and let me know if you get a busy tone.

Hello again,

Now we tested this in the following constellations:
- Call from A to B (B is busy) internal - busy tone works on A
- Call from A to B (B is busy) two different locations - busy tone works on A

So, there is a problem between client class and server?

It sounds like it. The thing I noticed was that the TWS GUI client application for me would give a busy signal most of the time, but a few times it would just break down the call. That usually means there is some kind of race condition. The Telephony Web Service has very little support of TSAPI. It is only meant to do a few simple things and doesn't do any call control or monitor stations. Avaya has a newer web service product called Agile Communication Environment, and it works with AES and CM as well. It has a wider range of capabilities. As such Telephony Web Service has remained almost untouched for several versions of software.

In your case you would need to collect the TWS logging, TSAPI logging, and the Communication Manager MST trace and open a ticket with devconnect. I would then submit the issue to our development team. If they determine a fix is needed then they would schedule it for a future release.

Upon further investigation, I found that the documentation supports the behavior you are seeing. InvalidPartyException indicates an invalid extension, or destination busy. In my case there were inconsistent results as I was encountering a race condition. This issue had already been submitted to development and is targeted for AE Services 6.3.3. To fix the inconsistency they are going with the way it was designed and that is to return InvalidPartyException when the destination is busy.

So what does it exactly mean? If they fix it in AES 6.3.3 it should work correctly (busy tone on caller phone) or only the InvalidPartyException works correct and no busy tone would be signalized.
If the busy signal works directly from AES Management Console, it should work in client software too?

We use an adaptation of the client software to work with our lotus notes client (Java). There are some buttons to dial a number from the internal telephone book. But we have some other dialing applications (internal software) too to provide CTI dialing in this applications.

So it's a bad solution to signal a popup with an error message for a busy tone. Furthermore we have to customize all our other applications and its different error message popup coding.

They are going to have it return the invalidPartyException as stated in the documentation. How it works on the AE Services web console is how TSAPI handles this scenario. AE Services handles this differently with the Telephony Web Service. I can really only speak to the functionality of AE Services and it's interfaces. In this case it is working as designed.

Ok, but how can it be, that it works on your tests with the TelSvcGuiClient sampleapp (Post from Jan 31 2013 8:35PM) and not on our tests? It's the same client software, the same AES Service and so on?

I also checked the Agile Communication Environment but this is an Notes 8.5 integration with sametime connect. We have Notes 8.5 only on Windows 7 machines and not on the XP machines (about 80% in the moment). So, it's not the solution too.