[Asterisk-Users] One way voice through NAT

Paul Lambert paul at digis.net
Wed Sep 3 08:46:06 MST 2003


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..
> 
> --
> ______________________________________________
> http://www.linuxmail.org/
> Now with e-mail forwarding for only US$5.95/yr
> 
> Powered by Outblaze
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users



More information about the asterisk-users mailing list