[asterisk-users] sip proxy

[Guy Gold]

On Sat,Mar 03 09:17:PM, K bharathan wrote:
>thru DSL connection the callers can hear only
> one way;asterisk pbx is behind NAT;


Greetings,
If the calls are esteblished,then SIP did its work.
The system may  RTP problems, NAT may or may not be the casue
for the issue. Defaults RTP ports for asterisk are UDP 10000-20000,
You may want to look into that direction.

Guy Gold