[Asterisk-Users] One way voice through NAT

[WipeOut .]

> 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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