[Asterisk-Users] Nat problem

[Damian Minkov]

I have the following situation.

My Asterisk Box is behind firewall ( for example 10.1.1.2 ) I have 
mapped 5060,10000-10010 and
in rtp.conf I have said this range of prots 10000-10010. I'm tring to 
dial a PSTN from another PC with Sip phone in internet with external ip.
I can hear the voice from the PSTN , but The Other Side can't hear me.
I ran Ethereal and so that all rtp packets  going from the calling phone 
are with destination 10.1.1.2.
What to do to configured it right ?

in Sip.conf
[general]
nat=yes
externip=213.x.x.x

[sipphone]
[damencho]
type=friend
username=damencho
host=dynamic
nat=yes
canreinvite=no