[Skumpic]

Hi i tried to configure my pstn->voip and my voip to pstn gateway functions but i find out some troubles.
I used spa3k web based utility to configure the spa3k with gossiptel.

First:Voip to pstn
Well when i call my inbound gossiptel number from my mobile, the spa3k after 5 second auto answer the phone call (GOOD).
Then i can heard the VOIP PIN Tone. I try to dial my pin (1234#)but the sipura did not recognize the number dialed!!!
I tried to call inbound gossiptel number with my mobile and also with x-lite.
Samre problem...

Second:PSTN to VOIP
Well i connected my spa3k to a friend pstn line.Now when i call the pstn line the sipura did not auto answer the phone call (i cannot heard the PSTN PIN tone).
Any suggestion??
Could it be the international control part of my PSTN page? (you can find it in attachment)

Ps.when i use the gw0 function in the dial plane in Line 1 page the call is correctly forwarded throught the pstn line.
If i put
PSTN Ring Thru Line 1:yes,
and then i call the pstn line the phone connected to the spa3k ring.
I'm in Italy
Thanks in advantage
Skumpic

[Skumpic]

The really strange thing is that when i try to make an inbound call to the pstn line for PSTN to VOIP gateway i can see in the INFO page of the sipura:
PSTN LINE STATUS
PSTN STATE: Ringing

So the sipura find out that the line is ringing (the status pass from idle to ringing)but it doesn't auto answer the incoming call....
I can't understand why!!!

Skumpic

[rizsher]

Skumpic, please post your actual config html page instead of the jpg screen shot.

[Skumpic]

Ok rizsher, here there are the 2 html page (pstn line page and regional page)
I hope that someone help me

Ps. When the pstn status is idle the pstn line voltage is -35 V
while when i pick up or make a call throught the pstn line the voltage is -6 V
Thanks
Skumpic

[Skumpic]

First problem solved
VOIP to PSTN call now work...
I made some changement in the PSTN LINE page
PROXY AND REGISTRATION
make call without reg = yes
answer call without reg = yes
VOIP TO PSTN GATEWAY SETUP
voip caller auth method = none
line 1 voip caller DP = none (this should mean no limitation...is it right?)
voip caller default DP = none

Now it works..finally!!!
Now i should resolve the pstn to voip gateway problem...
Skumpic

[Skumpic]

Ok also PSTN to VOIP works now.
It seems that with the default dial plan (XX.) the 2 gateways functions won't work (Bug??), while changing the 2 dial plan for PSTN to VOIP and VOIP to PSTN all works finely
See ya
Skumpic

[Skumpic]

I have received many pvt asking for my pstn tab configuration.
In attach you can find my config.
Hope this can help
Best Regards
Skumpic

[ritztech]

wheres the attachment ? im new to here .... thanks