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zerojay's Avatar
Posts: 2,669 | Thanked: 2,555 times | Joined on Apr 2007
#21
You can also wait for IT2008 instead of using the RTCOMM beta, if you'd like.
 
Posts: 14 | Thanked: 1 time | Joined on Aug 2006
#22
Is there no hope for the 770 under 2006?
 
iball's Avatar
Posts: 729 | Thanked: 19 times | Joined on Mar 2007
#23
But the Sofia SIP used in RTCOMM is still buggy. It won't work over a VPN connection for example.
And from what I've read in the bugzilla on it it's not going tobe fixed any time soon, not even in IT2008.
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R-R's Avatar
Posts: 739 | Thanked: 242 times | Joined on Sep 2007 @ Montreal
#24
Originally Posted by iball View Post
But the Sofia SIP used in RTCOMM is still buggy. It won't work over a VPN connection for example.
And from what I've read in the bugzilla on it it's not going tobe fixed any time soon, not even in IT2008.
Actually maximCT on #maemo said it worked over PPPoverSSH and over ssh -w (ssh tunnels)... But i don't have any more info for now.

I'm having the rtcomm client use the wlan0 ip instead of the tun0 IP! I'll see if i can fix this with the ip(8) command...
 
benny1967's Avatar
Posts: 3,790 | Thanked: 5,718 times | Joined on Mar 2006 @ Vienna, Austria
#25
Originally Posted by kcallis View Post
Is there no hope for the 770 under 2006?
+1 from me

Gizmo is a fine solution meanwhile now they allow alternative SIP-accounts to be used, but it doesnt integrate with the contacts and its - well, simply too much. If SIP could be integrated in the IM-framework on the 770, too, I'd love it.
 
IBICO's Avatar
Posts: 114 | Thanked: 7 times | Joined on Oct 2007 @ Norway / Tromsų
#26
When people talk about sip they usually talk about pc to pc, or to other lan devices. In Norway we got televoip that give voip over sip but where you call to other phones. The settinhs that televoip got is same as freeworldialup. So are there any client like gizmo that got same settings as FWD. did not find any settings like that in gizmo.
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benny1967's Avatar
Posts: 3,790 | Thanked: 5,718 times | Joined on Mar 2006 @ Vienna, Austria
#27
IBICO, which settings exactly are you talking about?

I use 2 sip-accounts with gizmo that allow me to call any landline/mobile phone, and i didnt have to tell gizmo more than account/server/password, IIRC.
 
IBICO's Avatar
Posts: 114 | Thanked: 7 times | Joined on Oct 2007 @ Norway / Tromsų
#28
Auth name, passwrd, server (sip.televoip.no), server port (5060), and maybe stun server with port. not sure about codec.
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