Voicemail/Forward is not triggered

[mbylica]

1. Brekeke Product Name and version:
PBX, 3.0.7.0 Pro

2. Java version:
jre 1.6.0_33

3. OS type and the version:
linux centos 5.5

4. UA (phone), gateway or other hardware/software involved:
Linksys SPAxxx, Siemens Gigaset, Zoiper softphone

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html :
Pattern 9

6. Your problem:

All the settings were set according to Brekeke PBX guides
- two accounts registered 100 and 101
- inbound section of 100 account settings has following data
--- Phone 1 = 100
--- Ringer Time (sec) =5
--- Fowarding Destination (Busy) (voicemail checked)
--- Fowarding Destination (No answer) (voicemail checked)
- ARS has default mediaserver_prefix route and one more for incoming and outgoing calls to/from ITSP

I can dial to UAs 100 and 101 from ITSP and make a phone calls from these UAs without any problem.
The issue i have is that i am unable to have call forwarded to VM, instead of this phone call is teared down after 60secs or so.
What is more i can dial 07*/08*100 and have VM prompts directly from 100.

Providing one more entry:
--- Other Fowarding Destinations* = 101
or
Fowarding Destination (No answer) = 101
or
Fowarding Destination (Busy) = 101
does not change anything. Still the call to 100 cannot be forwarded and is teared down.

Does anyone of you have similar situation or idea how to get out of this.

Thanks in advance.

[hope]

is the call transferred to 100 from auto attendnant?
if yes, is the call showing "talking" at sip server side "active sessions"?
if call from other extension but not ITSP, can call go to 100 vm?