I have a siemens gigaset s450IP working fine with freespeech, but when calling freespeech number from gismo like *399-freespeech_number the phone rings, but there is no sound both ways. Does any one have any clues what could be the problem?

The problem is your calling peer to peer and the gizmo client or gigaset is behind a NAT and does not support the 'direction:active' context in the SIP message.

Solution: Fix NAT traversal on both ends using STUN, port forwarding or SIP aware routers (built in SIP ALG)

My gigaset is using stun, as well ports from 5004 to 5082 are forwarded to the gigaset, what else I have todo on the gigaset end?

Thanks,
Alex

post your account number to the help desk. They can tell you if STUN is working properly.

Another thing to mention is, when I get a call from a gizmo account to my gizmo account registered with gigaset, then I have the sound but not the caller, in this case is clear that the problem is in the callers NAT/router. This might mean that my router is correctly configured. Correct me if I'm wrong.

Thanks,
Alex

Hmmm, freespeech echo test goes fine, landline to freespeech account is fine, freespeech to freespeech is fine, could it be that STUN is not working properly?

There is an application called winstun you can use to test if stun works behind your router. Google winstun should find it.

Thanks a lot, your thech support rocks

Thanks,
Alex

I found a solution for my problem.
I have an ADSL broadband with esatbt through a phoneline, so my network works throug an zyxel ADSL modem and a wireless router connected to it.

Initially my setting where like this:
1. modem was set in routing mode with PPPoE autentification
2. on the modem all ports where NATed to the routers IP
3. IPs between modem and router statically set
4. on the router I had an static IP distribution based on MAC values
5. port range SIP signaling (5004-5084) and RTP audio (8766-35000) NATed to SIP dvice IP

In this case SIP to SIP calls where fine, SIP to landline where fine, landline to SIP where fine. Gizmo to sip and SIP to gizmo was not working, NO sound.

Then I changed modem and router setting, points 1-3 accordingly:

1. modem set in bridge mode and RFC13xx (don't remember the numbers)
2. do not worry about this point
3. router is doing the PPPoE autentification now, so it is getting a real IP

So now winstun utility is happy, and gizmo to SIP and vice versa works fine.

YES ! for SIP to SIP to work correctly cascaded NAT (one home router behind another) causes lot's of headaches for the RTP ie. voice part