Ok thanks, so then I will use my statistics.
TargetStateEx still hides from me, and my IRD 8.1.300.22 refuses it as an unknown function - which IRD Version do I need to use it? Or is it an ECMAScript Function in Composer? (the migration to Orchestration Server / Composer is planned here, but will not happen for some months). But it's not a problem, currently, I can live with TargetState.
And because I've just got the attention of an expert... do you have any idea why this function sequence ignores my timeout setting? In the SIP Server manual, they write that after-routing-timeout is overridable with this Extension. divert-on-ringing should be false for this timeout to work, so I added it after my first tries went wrong (but the option is set to false in SIP Server application anyway...). The log shows me that the call is not diverted before ringing, so this setting is ok.
[code]
ExtensionUpdate['divert-on-ringing', 'false']
ExtensionUpdate['after-routing-timeout',7]
TRoute[var_directTarget,var_directLocation,RouteTypeUnknown,'']
[/code]
But it doesn't work, my TRoute receives the timeout after 20s, which is the SIP Server application setting. I also tried RouteTypeDefault, with no effect. Any idea what I am missing?
Admittedly, nas-private is false on SIP Server application level, but I set it to true for my agent login and for my DN. Even for the Routingpoint where the strategy runs (ok, that was silly). But nas-private shouldn't matter, because the strategy runs on a routing point and on entry, I see the Extension BusinessCall=1. And when I answer and release the call, the agent is set to after-call-work, which should be proof that the call is considered as business call.
Obviously, I am missing something. But what?
SIP Server is Version 8.1.102.55, Switch is Open Scape Voice from Siemens (or Unify) - but that's a SIP Server extension, so the switch shouldn't matter.
Ah yes, one addition: I could use divert-on-ringing=true and set the NO-ANSWER- Extensions, which works. For a given value of "works". Because I can get the SIP Server to do nothing (that is: the call stays where it's been routed), release the call (kicking the caller out) or to requeue it to my routingpoint (causing my strategy to start over). Starting over means that I need to detect this, and cache some computation values in attached data. This is a complication that I'd like to avoid. And I don't know what InfoMart makes of those requeuings.
Thanks for any insights.
Rolf