[maemo-developers] [maemo-developers] VoIP through SIP

[From: Jaime Ruiz Frontera]

On Fri, 25 Aug 2006 20:00:53 +0200
Johannes Eickhold <jeickhold at gmx.de> wrote:

> I was able to replicate your segfault using my own SIP provider. I made
> a new release of the package which fixes the problem. Should now work
> for incoming and outgoing calls even if you are behind a NAT. (hint: my
> provider provides a STUN server which was automatically found and used
> by sofsip_cli) Get the package from the link above.
> 
> Greets,
>   Jonek.

Amazing! It just worked! I could call home for free! 
I felt like Alexander Graham Bell in his first telephone call :)

Thanks a lot for this great work.
Jaime


-- 
Jaime Ruiz Frontera

e-mail: jaime AT cauterized.net
jabber: jaime AT zgzjabber.ath.cx
sip   : jaime AT ekiga.net
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