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
_______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
