Route call after disconnect

[Tambo]

Hi

[color=red]For these purposes actually exists option "after-call-divert-destination" on SIP server application object. Try to use it.[/color]

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

[Kubig]

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.

[Tambo]

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

[Kubig]

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 ;-)

[Tambo]

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

[Kubig]

I did a quick test and all works fine on RP object level. I do now know how you have configured it, but in annex must be the "TServer" section, under this section must be option "after-call-divert-destination" with any valid DN as a value. That's all.

[Tambo]

Hi Kubig,

there i think may be the problem..........i didn't have a TServer section in my Annex tab of my RP, but i have now created one and still no joy :-(

i think i am being over ruled by other config ?!?!

really appreciate all of your help

Tambo

[cavagnaro]

Check your logs, if applied you should see it there

[Tambo]

Hi Cav, logs don't show the divert they show call disconnect then call delete.

Would anything in call cleanup etc stop this working

[Grand_Master]

Add the option and see what happens in the logs.

[Tambo]

rookie mistake.....sorry lads

SIP server 8.1.000.86 is the correct version for this to work but we're on 8.1.000.81  ?!?!?!?

going to upgrade !

cheers for all your help

Tambo