Re: SB Error + VOIP routing

["darrenp_lock" <darrenlock@xxxxxxx>]

James,

I have version 0.1.0.194 of xAPSwitchBoard installed, which is
running fine. However, if I extract the exe in the zip below over it
then run the program, nothing happens. The busy cursor appears for a
fraction of a second and then nothing. Doesn't even survie long
enough to appear in task manager.

Any thoughts?

Rgds, Darren.

--- In xap_automation@xxxxxxx, James Traynor <james@...>
wrote:
>
>
> http://mi4.biz/files/testing/20070211switchboard.zip
>
> You can now set the network list and the list of networks that have
the
> disconnect option. I have added a 'Integration' tab which details
how to
> link switchboard to various things including the asterisk callerID
http
> serivce., still got to add more to it though!
> Now if a call comes in and the number is matched against the
database
> and if that contact's number has a network set then that network is
used
> to return a call not the one from the incoming message.
>
> James
>
> James Traynor wrote:
> > Gregg Liming wrote:
> >> Quoting James Traynor (2/8/07 7:37 PM):
> >>
> >>
> >>> Gregg,
> >>> This Switchboard also contains outgoing network selection
per
contact
> >>> per number we talked about. It still needs some work but
it
should help
> >>> with asterisk routing.
> >>>
> >>
> >> This looks really good!  A couple of questions/issues though:
> >>
> >> 1) Now, I'm seeing an automatic "Disconnecting in 7
seconds"
message as
> >> soon as the dial goes through.  This appears to be new.  Does
axc need
> >> to send a CTI.cmd w/ a CTI.cmd Offhook message to force it to
continue?
> >>
> > This was a feature added a while back to pstn calls to allow a
dial to
> > be aborted. I'm just finishing of the update and that has a csv
list
> > of network types that this appears with so you can change teh
behavior
> > as needed.
> > You can also customise the network drop down list on the phone
number
> > screen.
> >
> >>  FWIW: axc is now using the strategy that we discussed where
the
VPSTN
> >> line remains declared free since it can create an indefinite
number of
> >> trunked calls.  I'm wondering if the previous line state
change
kept the
> >> disconnect message from appearing--especially since the
original
> >> behavior persists for (V)PSTN calls.
> > I think it was just the fact of switching from Vpstn which it
didn't
> > appear from to pstn where it does.
> >
> >> 2) This is truly a "nit", but if initiating a dial
from the call
log of
> >> a number that is now associated w/ a contact, it seems that
SB
still
> >> assumes the line type to be PSTN rather than propagate the
line
type
> >> that is now selectable in the contact record.  In contrast,
dialing via
> >> the contact record is perfect.
> >>
> > A dial from the call log uses whatever network was reported when
the
> > call came in, I assumed a returned call would go back along the
same
> > network it came in on. I will look at using the network from the
> > contacts db if it's set.
> >
> > As a side note I was playing with the new trixbox ( vmware
version)
> > and noticed a callerID lookup function. It does a DB/HTTP lookup
to
> > collect a caller's name. Anyway I've added a url in switchboard
where
> > you can point this too so you now get callerID names directly
back
> > into * and then on tot he phones. This is just to provide name
info
> > back to asterisk, it doesn't effect any of the call logging or
> > tracking that is based on the axc plugin.
> >
> > Should be a update out ina  few hours
> >
> > James
> >
>