[Asterisk-Users] Asterisk in parallel with PSTN [OT]

[Rich Adamson]

> > >I've got a configuration with PSTN line connected to FXO
> > >on TDM400P ringing through to a phone connected on a
> > >Sipura SPA-3000.  The phone *does* ring before the
> > >caller-id is available.  In fact, it shoes some 
> > >alternate message like "waiting for caller id info"
> > >right after the first ring and then changes to
> > >the real caller-id after the 2nd ring.
> > >
> > >-Dorn
> > 
> > I've always wondered if certain IP (regardless of proto) phones could do 
> > the same?  Basically initiate the call with fake callerid info and then 
> > send an 'update' packet later to inform the phone of the new callerid? 
> > Is this possible - even if it is only supported on certain phones?
> > 
> > If this is possible, then we could modify * to allow the dialplan to 
> > (optionally) start before callerid is received and then update the 
> > ${CALLERID} variable(s) once the information is available.  There are 
> > situations where this is VERY desirable (obviously this only applies to 
> > POTS though).
> >
> Seems like something similar must be going on in my setup,
> because * is clearly taking the inbound call from the 
> TDM400P/FXO and ringing it through to the Sipura FXS port
> before the caller-id info is available.

The zapata.conf entry for the channel will need something like:
 immediate=no
 usecallerid=yes
If you have an analog phone on that same pstn line, you should notice
that * won't ring the internal sip phones until after the second pstn
ring. The CallerID is simply passed to the sip phone without any
special variables, etc.