SPA3102 and Freepbx 2.8.1.4 under Pbx in a flash with asterisk 1.8.5

Hello,

I m running Last Piaf (PBX IN A FLASH) with Freepbx 2.8.1.4 on local Dell computer. My previous installation was ( PIAF purple - asterisk 1.8 and freepbx 2.5.6.2) my spa3102 was ok.

today, when upgrading with last piaf to freepbx to 2.8.1.4, my spa3102 has the 4 green leds ok (trunk an extension seem to be well connected and registred) but when is not able to handle incomming calls. Asterisk says “the number dialed is not in service, please verify it and try again” ???.

Thank you for your help …

extension “610”

[610]
deny=0.0.0.0/0.0.0.0
disallow=all
secret=xxxxxxx
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=no
port=5060
qualify=no
callgroup=
pickupgroup=
allow=alaw
dial=SIP/610
[email protected]
permit=192.168.1.4/255.255.255.0
callerid=device <610>
callcounter=yes
faxdetect=no

trunk “pstn”

[pstn]
disallow=all
allow=alaw
canreinvite=no
context=from-pstn
dtmfmode=rfc2833
host=dynamic
call-limit=1
nat=never
port=5061
qualify=yes
secret=xxxxxxx
type=friend
username=pstn

Registration :
Trunk : pstn/pstn 192.168.1.4 (spa’s ip) D 5061 OK (10 ms)
peer : 610/610 192.168.1.4 D A 5060 Unmonitored

Hi,

Your trunk config looks good a first glance… remember that the linksys use 2 stage dailing when a call from the PSTN comes in.

Look at the setting : “PSTN Caller Default DP:” x
Then look at the corresponding “Dial Plan x:”

What do you have entered there, tis is the VoIP number that will be dialed whe a call on PSTN comes in, enter this in that Dialplan Field : (S0<:610>)

This will Dial DID 610 on the PBX, wich will fall trough to an extension as you probably had in 2.5 (no experience with that version)

Otherwise an incoming log would help.

Regards,
Richard

Have you seen or attempted to follow the instructions here:

http://www.freepbx.org/support/documentation/howtos/howto-linksys-spa-3102-sipura-spa-3000-freepbx