Sipura SPA1001 setup problems

Asterisk Asterisk Support
1

I’m currently setting up a Sipura SPA1001 at my home office to connect to an asterisk server over the internet. The * will actually provide a gateway connection to an analog extension on our Nortel KSU through a FXO port (Digium TDM).

From an inbound call to the Nortel system, as well as dialing the extension from in the office, the * server correctly routes the call to the Sipura adapter, and the phone rings. But there is no audio on either end. Also, * server has to be restarted before another attempt can be made.

Both sides of the connection are behind NAT.

Any suggestions?

2

After researching posts here, and suggested reading from voip-info, I was able to get it going. I tried quite a few different things, but here is a list of the things that are in place now:

1)I setup STUN support on the Sipura device using a public STUN server

2)used port forwarding on the * server end of the connection.

3)set “Nat=yes” in sip.conf

4)set “externip” paramater in sip.conf

3

[Hi,

I recommend you not to use STUN server.
Fill on Sipura field “Ext IP” of your NAT and “Subt via Addr”

There is no voice at both ends because of NAT. It doesn’t pass through rtp
Do you have access to NAT?

Regadrs, Dim