What router model do you have? Perhaps that would help. It would also help if you tried on some other connections. Although I have to say that my VOIP dreams have been curtailed by corporate greed. (The whole AT&T debacle, Gizmo now, limitations with this and that)

EDIT: By that I just mean, it reminds me of "Who Killed the Electric Car?"

It's a Westell A90 from Verizon.

I assume you already checked these two guides (not sure which revision you have)

http://www.portforward.com/english/r...mo_Project.htm

and

http://www.portforward.com/english/r...mo_Project.htm

(look the same to me though)

But I'm going to have to say that it's not just your router. I had similar issues with Gizmo.

Yeah, I found those instructions, but they don't even come close to matching my router's forwarding admin pages... very different model version and/or firmware I guess.

Ok, I confirmed it: my home network is the culprit. I tested the N810 at my job and could call out using Gizmo, hearing fine at both ends. So somehow I have to figure my firewall issues out.

I could not call to the tablet though, confirming what was said about Google clamping down it appears. Hopefully I can get this working using sipgate.

has anyone figured out how to confirm/associate a google voice number with an ipkall or sipgate # through the tablet? or any other way? officially each gvoice number is supposed to be linked to a "physical" phone, and i already have another google voice number associated with my gizmo sip number on the tablet, but would like to keep and use the extra one i just got for another email acct. for shopping etc.

Sick. Usually it means incoming on WAN. Can you put it in bridge mode and use some embedded device over which you have real control to do the hard work? Or make it forward all ports to/from this embedded device? The thing to look out for is that it has to be able to process your FIOS speed (and anything additional you might want to add on the box, such as Asterisk or Sipproxy).

...

Are you using a sipproxy? Do you use any other SIP in your network? Do you have UPnP enabled?

God's ways are mysterious...

VoIP is usually illegal on Dutch 3G networks, and some providers do block it. Try to create a VPN.

BTW your range doesn't include RTP. Those are used too. On your SIP client or PBX you can define which ports your SIP application wants to use and use UPnP, or manual forwarding.

Try to run packet sniffer on N810 and you probably see it tries default RTP ports which you didn't mention earlier. Try to forward these. Usually, it tries in RTP port IANA assigned the RTP_port and if that one not work tries RTP_port+1 and so on. Additionally, a SIP proxy can help with such problems, it is a STUN server.

You can try this public STUN server too: stun.counterpath.net

Not all SIP providers support SMS send over VoIP, and I'm not sure how the billing works. If I understand correct one connects to a SMSC then, which does the dirty work?

PS: I can give you temporary access to Dutch SIP server & STUN server including PSTN phone number with country code +031. Optionally routed via OpenVPN. PM if interested.

[EDIT]Merge posts[/EDIT]

calls from the regular gv website currently will still ring gizmo, while calls from the gv mobile site, which i had been using, don't anymore. anyone else experience this?

Yes, I have had that problem too. I think that's a major reason why Dialcentral doesn't work with gizmo either.

I'd had that problem, but managed to get the mobile site working by changing the settings and selecting my gizmo number (even though it was already selected). It still didn't fix dialcentral, however.