Hi All,
I have been asked to provide routing for a GVP app after calls have finished. I want to have an agent talk to customer, then at end of call disconnect from customer - this then would transfer customer to my GVP app, the app would play then customer would get cut off.
I have tried ‘DeliverCall’ with a DN number and get error message in logs saying 0008 can’t use because of using ‘Routed’
am i barking up the wrong tree with this one?
any ideas gratefully received
T
For these purposes actually exists option “after-call-divert-destination” on SIP server application object. Try to use it.
PS: Disconnect must be from non-customer site of course 
Hi there!
can you explain - how the GVP can play something to customer if you whant router the call after the custoemr is disconnected?! you want to create an outbound call?
WBR Tim
I guess he wants kinda a survey maybe?
You can if you have 2 legs on GVP stuck to the call, the transfer done to the agent can’t be single transfer but more like a conference, so when agent releases the call it will keep on GVP.
Another option is that on your softphone instead of Release the call you program it to do a Transfer to GVP.
No automatic way to do this…I spent almost 2 years arguing with a customer who said that was “easy” to do and didn’t want to do a softphone modification…those days I just wanted to kill him…lol
Hi guys,
Thanks for the input
Kubig - may have to leave this as last resort as don’t have full test environment and would need to get support to test and quote for it 
Timur - it’s just as Cav says really
Cav - going to get me one of those metal helmets Magneto wears in Xmen ;D yes it’s a survey app which I have working perfectly but need to figure out now the customer ‘opting in’ at the start of the call to delivering to the app. Looks like I’ll have to try the conference way of doing it as once customer opts in then shouldn’t matter who disconnects.
Thanks guys
T
I do not know, why you do not use mentioned options, which exists for these purposes- check the description of the option. It is the most easist way how to achieve what you want, one option, no “workaround” on DN level or difficult routing.
Totaly agree with Kubig. If u just need to route existing call from customer after disconnect from agent side - use the after-call-divert-destination options to route call on “survey” application. It’s prety simple and almost work in all configuration with SIP server that i ever seen.
Kubig, Timur
thanks for this, i will now try down this direction and see what happens
cheers
T
Well that will work if agents are SIP agents only…if they are on another PBX then no way to do such as suggested
We modified GAD disconnect button script to transfer customers to post-call survey RP, but Genesys does not like it 
Haha yeah, we did that too and answer “We don’t support that…” whatever…lol
Actually it is possible - if You pass call through SIPS and than deliver it to PBX SIPS option is still working correctly
Well…using 2 SIP channels…the same as a consult call
Not exactly - SIP channels are less resource consuming than two SIP/RTP channels during consult call
Hi,
I did in my environment (GVP/SIP 8.1 version), after agent talk with customer then first Agent need to disconnect the call then from softphone you can route a call to another DID / trunk number then you can use GVP to play else you ca route a call to SIP RP so in the strategy you can call play application to play ( for this also you need VXML application).
Thanks.
T
Hi
For these purposes actually exists option “after-call-divert-destination” on SIP server application object. Try to use it.
Kubig, i have put this option in the SIP server and set it to true. In the RP DN i’ve also put it with new DN number to divert to and can’t make it work ?!?! i’ve even tried puting the new DN into the Default DN tab of the RP ?!?
any ideas where i’m going wrong?
cheers
Please, read the description of this option and all will be clearly for you. The option has not “true” or “false” as valid values. Valid values is any valid DN number.
Hi Kubig,
I kind of thought that but can’t set that value in ‘live’ SIP server as haven’t tested the opt in opt out part. Surely if i do this then all calls will be affected??
cheers
As I know, you can set this option on application, RP or RequestRouteCall attributes level. So, try to configure it on RP level and make a test call. Option must be under TServer section in annex of specific RP object in CME.
All is described in deploy guide - read the doc first 
Hi Kubig,
yes been over it several times and it’s not worked hence the reason i thought it would be a global seeting that unfortunately i can’t test in ‘live’ environment
Call Divert Destination
SIP Server supports routing the caller to a specific destination when, after an
initial leg of the call is completed, only the caller remains on the line. For
example, this feature could be used to route the caller to a post-call survey.
Feature Configuration
To enable this feature, configure the DN-level option after-call-divertdestination
on the Routing Point DN. You can also enable this feature by
passing the after-call-divert-destination parameter in the Extensions
attribute of a TRouteCall request. Parameters passed in the Extensions attribute
override the value of the configured option.
I’ve tried different variations of the DN number in the RP annex tab with no joy
cheers