[Asterisk-Users] AudioCodes MP108 8-Port FXO Analog Gateway (SIP)

[Ing. Angel Gomez Garcia]

Hello Ernest.
I'm setting up two * boxes using mp108 ( FXO and FXS ), and it is 
working fine. The only issue with the FXO box is that it does not 
support remote disconnect supervision so you have to make sure that the 
reorder tone ( wich is used for disconnect ) is adequate for your country.

I have this in sip.conf:
------------------------------

[mp108out]
type=friend
host=x.x.x.x            ; <-- Fixed ip assigned to the mp108
dtmfmode=inband

[mp108in]                 ; <-- mp108 configured to register with user 
mp108in
type=friend
host=dynamic
dtmfmode=inband
context=inbound

------------------------------------

And this in extensions.conf
------------------------------------
[turnklongdistance]
exten => _91NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@mp108out)
exten => _91NXXNXXXXXX,2,Congestion()
exten => _91NXXNXXXXXX,102,Congestion()

[longdistance]
ignorepat => 9
include => trunklongdistance
...
...

[inbound]
exten => s,1,Wait(2)
exten => s,2,Answer()
.....                                       ; Basically your ivr main menu

exten => 100,1,Goto(s|1)

exten => 200,1,Dial(ZAP/2-1)
....                                                ; Handling of exceptions

...                                                   ; More extensions

-----------------------------------------------

on inbound calls you have to configure the mp108 to forward incoming 
calls to extension 100.
on outbound calls you have to configure the mp108 to One step dialing.

choose the order of your codecs and i think thats it.

Good luck.