[Rtcomm] [Rtcomm] [Bug 6171] Cannot establish received call via SIP with Google Voice

[From: bugzilla-daemon at maemo.org]

https://bugs.maemo.org/show_bug.cgi?id=6171





------- Comment #3 from kimitake at gmail.com  2009-11-17 23:34 GMT+3 -------
(In reply to comment #2)
> Wait, this one is different: the user doesn't even have a chance to use DTMF
> before the call is routed elsewhere.
I think this bug is same as bug 5505.
In my case, Google Voice server calls my SIP number and
I can connect to the service first.
The server requires "1" button pushing, it means I need send DTMF tone.

And if the service gets the tone, it calls the peer.

But I cannot send the DTMF tone to the Google Voice service,
so it does not call the peer and goes to voice mail instead.

> I suspect the real problem is in SIPphone connection. Please open your SIP
> account settings and make sure the transport is set to UDP.
> Please also fill in the STUN server (uncheck the autodetect checkbox) to e.g.
> "stun01.sipphone.com".
> Please report if the problem persists with these settings.
I just tried to change the STUN server but it is same.


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