We own a VS-GW2120 with 8 GSM-modules (32 ports). We use it as a SIP server with registered endpoints.
We routed all incoming calls to all gsm ports to SIP endpoints.
The hardware is behind NAT.
Everything works find for the first gsm module. So ports gsm1.X work find.
However, call to any port on any module above 1 (gsm2.X, gsm3.X, etc.) gives us a one-way voice. RDP goes to the internal endpoint's IP address and never reaches it.
We tried everything: STUN, forced NAT, disabled NAT and so on. Tried to add and remove Internal networks in the Advanced setteing. Nothing helps: SIP user can't hear the GSM user. GSM user can hear SIP user.
So the same SIP endpoint receives calls from the first gms module (gsm1.X ports) without any problems. Calls from any other gsm module give one-way voice (from SIP to GSM), but doesn't work in the opposite way (from GSM to SIP).
I suppose that might be caused by differnt IP address used for sending RTP traffic for modules starting from 2.
For example, i have LAN IP 192.168.10.222 as main LAN IP. And all modules are in cluster and have their own internal IP addresses. What IP address will be used to send RTP from, for the second module, for example?
I just logged via SSH to OpenVox and got good and bad calls information (sip show channels and the sip show channel for each channel)
Good one (from the main module gsm1) has only one peer (directly to endpoing) and the bad call had two peers, one from Asterisk to the endpoint and the second is from ASterisk to another Asterisk on he gms2 module.
I suppose IP addresses of the second gsm module somehow related to that, but how?
Sorry for the trouble. The internal ip of slave boards is used to communication with master board(board 1). When communication with sip server, all the modules will use ip of master board. Please prepare remote access for checking. I will help you to figure it out. For convenient contact, please add me as your skype friend.
Best Regards,
Linda
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