Why is there no documentation for developing for Genesys SIP Server

[victor]

Hi, all,

am I the only one who is getting frustrated with lack of information on how to program with Genesys API's?

I have been playing around with Genesys SIP Server and documentation about how to program for it is no where in sight. Does anyone have any reference on how to develop a SIP phone using Genesys CTI? I have looked at .Net GAD developer guide and ActiveX one and there is no word on it. I think it is really strange, because by using Genesys SIP, you get both CTI and SIP controls that are handled by T-Server, and not providing a well documented developer guide for it defeats the purpose of using it, because if we have to still rely on VAIL, TAPI or RTC for SIP-related control and use Active X for CTI, I might as well be using Cisco.

(Venting my frustration)

Vic

[René]

Hi Vic,

First I have to say that I'm sharing your frustration about Genesys SDK's. I'm leading development of agent's application based on .NET Toolkit (handling emails only) and it's horrible. Sometimes it seems to me to be worse that development with ActiveX/COM components for ICS 6.5...

Genesys .NET Toolkit does include a sample SIP Client application. It's based on .NET Toolkit AIL Services (communication with TServer) and Microsoft RTC stack (SIP handling). Maybe that information could be helpful for you.

Generally I would say you can use any existing library for communication with TServer (ActiveX, Java, Voice Platform SDK etc.) and any existing library for SIP protocol because these two protocols are independent even used in one application. And to be honest I wouldn't expect Genesys delivering any SIP library out-of-box...

René

[victor]

Hi, René (ALT-0233  ),

thank you for your reply.

Yes, it does have it, but I simply do not see a reason why they would not have a unified SIP/CTI library, because to me it seems like it is a very logical step in expanding Genesys CTI signalling. [b][color=red]Especially if they are planning on people using their SIP Server[/color][/b]

I am also using RTC and Genesys ActiveX, but you know, sometimes, it is really frustrating, because basic things refuse to work. Develping an attended transfer provides to be a total nightmare!

Here how it is according to Cisco:
[attach=1]

And interestingly enough, it is not working like that when we are attempting to do it with RTC 1.3 and SIP Server 7.2. I am sure that probably we are not having the right tag or CallID or SDP parameter, but, what is the point of using TTransferInit for example, if we still need to handle INVITE, SDP and the rest of the commands using a Microsoft-based library? We might as well start the whole call control routing (including transfer) using Microsoft RTC and just use Genesys CTI for popup and Agent state control. And if we are to do that, I might as well be coding for Asterisk or whatever other SIP server is out there. If Genesys is really serious about its SIP line, it should get its act together and come up with a complete solution and not ask us to put together bits and pieces of technologies from three different vendors. I feel like I am back using Aspect CTI which was all about scripting and required deep knowledge of SQL, Java Script, and pretty much made you feel like you were a Web developer using Notepad to develop a company homepage!