I have troubles to use Gizmo Area 775 free service with N800. Incoming calls ring on Nokia, I have a popup window to accept or deny a call but that's it - after I click on accept button, I see a "chat"(?) window and greeting message starts to play immediately. I still hear another party voice, but he does not hear me. Period. Looks like outcoming SIP traffic gets cut somewhere in the middle. To make calls to N800, I use my home Comcast phone. I tried to forward ports (UDP 5004,5005, 64064) to N800 directly in router setup, but it doesnot help. As I see here, the problem is in outgoing traffic and I honestly do not know how can I fix it since my router does not block anything that is coming out.
Any thoughts that might help here?
Do you have another computer or SIP hardware phone in your private network? I noticed that if I leave my Grandstream VoIP adapter connected in my network I see the same problem... Only if I disconnect the Grandstream adapter the call connects properly to N800.
I have 4-5 PC's in the network, all under XPs, no domain, Workgroup only. And no SIP hardware. Somewhere in internet somebody mention DMZ server settings in router needs to be set, but I have no idea what is it and to what value I need to set it to.
What Router do you have? It might be worth checking if there are newer firmware for the router... that might solve your problem.
Other than that, you might try to disconnect all other computers from your network so that the N800 is the only device. If that works, then it should be just a matter how the set up routing properly. If not, then setting your N800 into the DMZ might be worth a try.
EDIT: You mentioned that you use Comcast phone... It may want to use the same ports than Gizmo... and that may have something to do with your problems...
Can I change outcoming SIP port to something else in my Gizmo client?
I could not find these settings.
I don't think there is...
But you can use the built in SIP client to connect to Gizmo VoIP server. It is in the control panel accounts... and define new SIP account. Just put these settings:
Proxy: proxy01.sipphone.com
STUN: stun01.sipphone.com, port 3478
The user ID is either your Gizmoproject login ID or you can also use the Sipphone "phone" number... both will work.
In the SIP client you can define the port number. The default is 5060.
