Genesys CTI User Forum

Genesys CTI User Forum => Genesys CTI Technical Discussion => Topic started by: PFCCWA on February 27, 2014, 11:16:41 AM

Title: Divert callers to GVP upon disconnection
Post by: PFCCWA on February 27, 2014, 11:16:41 AM

hello,

We want to divert callers into a GVP at the end of each call (this will play vxml application asking for DTMF responses to questions).
What is the recommended way to do this?

I think we (agents) cannot simply disconnect the call and have routing strategy send to vxml?
Do we need a custom button in GAD, that disconnects call with agent and diverts to GVP?
Any other way which does not require customization?

much appreciated.
thanks.

Title: Re: Divert callers to GVP upon disconnection
Post by: Kubig on February 27, 2014, 12:12:27 PM

For this purpose is on SIP server option "after-call-divert-destination"

Title: Re: Divert callers to GVP upon disconnection
Post by: PFCCWA on February 27, 2014, 03:23:27 PM

thank you.

defining this option in route point dn or sipserver applications would not work however using function ExtensionAttach in ird routing strategy did work.

Title: Re: Divert callers to GVP upon disconnection
Post by: Kubig on February 28, 2014, 08:07:23 AM

What version of SIP server are you using? I have configured it on application level, ie. SIP server, and it works properly.

Title: Re: Divert callers to GVP upon disconnection
Post by: cavagnaro on February 28, 2014, 02:10:14 PM

This works only if agents are SIP too, right?

Title: Re: Divert callers to GVP upon disconnection
Post by: Kubig on March 03, 2014, 08:37:24 AM

I think no or documentation does not describe something like that

Title: Re: Divert callers to GVP upon disconnection
Post by: cavagnaro on March 05, 2014, 02:16:34 AM

So this means SIP Server keeps the leg when doing a transfer to an agent (Any premise TServer)? Will consume then 2 ports?

Title: Re: Divert callers to GVP upon disconnection
Post by: Kubig on March 05, 2014, 07:09:05 AM

I have never seen similiar deployment, where this option is used...so cannot say how it works in real :-) Just always use in pure SIP env