Just seen on simoniito.blogspot.com that a beta version of the next Connection Manager software has been released.

Main Site is here :-

rtcomm.garage.maemo.org

Its got a working SIP implementation and its working fine with my Asterisk server.

One can never have too many VOIP clients!

If you've got an updated N800 give it a try.

its good to search ...

http://www.internettablettalk.com/fo...ead.php?t=7761

A little late.

How do I get an "SIP number" (or account)? I have an account with Gizmo Project, can I use it?

Also, (probably a stoopid question), is Vonage some sort of SIP?

Yes. Look at the rtcomm site for information on it (sipphone.com).

Don't know.

There are a lot of telephone numbers around the world that you can call to be connected to the SIP networks. One of the benefits of SIP being a standard.
http://www.sipbroker.com/sipbroker/action/pstnNumbers

I can receive calls using this software but I cannot place calls. When I dial the N800 "sounds" the ring tone but the receving number doesn't ring. Any advice?

Vonage is SIP, though they use weird port numbers. They will never tell you your configuration, though, so you would have to unlock your existing Vonage adapter so that you can see the settings. I know with a PAP2v1, PAP2v2, or an RTP it's possible to unlock it and still use the adapter with Vonage. Other adapters may allow you to unlock them, but not in a way that allows you to see your settings so you can unlock them.

With a PAP2v1 you'll have to call Vonage tech support and make up some problem, like frequent call drops or something and say you want to downgrade. Say your friend got a downgrade, and his adapter works great at your house so you think the latest firmware doesn't like your router or something lame. Instructions for all of them are on the bargainshare.com forums or other places accessible with Google.

I'll try my Vonage account with the new Connection Manager later this week and see if it works

Edit: Doesn't appear to. The connection manager says it's connected to Vonage, but calls give a server not found (or something) error. Oh, well.. that's why I'm switching to Vitelity anyway

To ring landlines / old phones you will need to buy some Gizmo credit.

To test your callout capability , try calling the echo test user (gizmo username is echo) and see if that works.

To follow up, I have access to a sip server and have a phone number...we own a softswitch. We "discovered" that it appears to be a problem due to NAT. WE are going to fiddle with the sip server tomorrow to try and solve the problem.