[Asterisk-Users] One way voice through NAT

[Paul Lambert]

Thanks, that worked.

"WipeOut ." wrote:
> 
> > I'm connecting and can place calls to and from my SIP phone that is
> > behind a firewall, can hear audio from the SIP on the PSTN line but
> > can't hear audio on the SIP phone from the PSTN line. Anyone else
> > experience this?
> 
> Try adding "nat=yes" in the config for that UA in the sip.conf..
> 
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