I'm a very newbye of asterisk and visdn :-)
I have connected an asterisk box to S0 bus of a traditional PBX,
now i can call from a sip phone an internal number of PBX but
if i call a sip phone on the traditional headset i can't hear ringing
and if i accept the call on the sip phone all work fine.
Here is the my visdn.conf
[general]
[global]
network_role = private
tones_option = yes
outbound_called_ton = unknown
force_outbound_cli =
force_outbound_cli_ton = no
clir_mode = default_off
cli_rewriting = No
national_prefix = 0
international_prefix = 00
network_specific_prefix =
subscriber_prefix =
abbreviated_prefix =
overlap_sending = Yes
overlap_receiving = No
autorelease_dlc = 10
call_bumping = No
[visdn0]
network_role = private
context = default
tones_option = Yes
outbound_called_ton = unknown
force_outbound_cli =
force_outbound_cli_ton = no
clip_default_name = Centralino
clip_default_number = 291
clip_numbers = _29Z
clir_mode = default_off
overlap_sending = No
overlap_receiving = Yes
Here is the relevant part of extensions.conf
exten => 291,1,Dial(SIP/291,20,rt)
exten => 291,hint,SIP/291
exten => 292,1,Dial(SIP/292,20,rt)
exten => 292,hint,SIP/292
exten => _X.,1,Dial(VISDN/visdn0/${EXTEN})
Thanks, Andrea
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