[Asterisk-Users] X100P configuration Problem

[Daniel Concepcion]

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