asterisk users - Nov 2003 - X100P configuration Problem

Hi People,

I have the following scenario:

PSTN via Ibercom - 3 x X100P -  Asterisk - Sip phones

Ibercom = A product of Telefonica in Spain, interconnecting with old Ericsson 
equipment buildings of the same company via PRI and also connecting with PSTN 
via PRI.

My problem is that when I have an entry call via X100P and I redirect this 
call to the voicemail or conference room. The caller give the msg and when 
hang up the voice mail save 180s of busy tone until timeout and hangup the 
zap channel or i see the busy tone in conference room until the call timeout.  

If i answer the call in the Sip phones when I hangup the Zap channel also 
hangup correctly.

I think that I have correctly the indications.conf.
Someone have any similar issue or know some workaround? 


[es]
description = Spain
ringcadance = 1500,3000
dial = 425
busy = 425/250,0/250
ring = 425/1500,0/3000
congestion = 425/200,0/200,425/200,0/200,425/200,0/600


regards,

Daniel

On Sat, Nov 22, 2003 at 06:05:06PM +0100, Daniel Concepcion
wrote:
> 
> My problem is that when I have an entry call via X100P and I redirect this 
> call to the voicemail or conference room. The caller give the msg and when 
> hang up the voice mail save 180s of busy tone until timeout and hangup the 
> zap channel or i see the busy tone in conference room until the call
timeout.
I have seen similar things. I believe it has to do with the
difficulty in detecting when the far end has hung up in the
absence of digital signalling information. With analogue
interfaces like the X100P usually they will detect a reversal
in polarity and take that to mean that the far end has hung up,
and then hang up themselves. 

But this is with telco provisioned POTS loops.

I suspect your Ibercom box is not reversing the polarity,
so the X100P has no way to tell that the call is gone.
Perhaps it is possible to configure the Ibercom to make
it do the right thing?

-w