Urgent Help Needed on NAT

  • We have VoIP Switch and Asterisk Box behind pfsense firewall. pfSense have Live IP address and when our carrier send SIP call, first of all call comes on firewall then goes to VoIPSwitch and then Asterisk. pfSense have live IP address 118.XXX.XXX.XXX and its LAN interface have VoIPSwitch IP address is and Asterisk BOX IP address is Call connected successfully and carrier can hear from our side but we are unable to hear him. We have define NAT from WAN interface to against ports 5060 and for RTP 6000 to 30000. I think RTP packets not come from carrier side but from our side these are ok. Please guide us if our configuration is not correct.

    Best Regards,

  • Go to NAT section, and under outbound part, enable AON and in the rule that appears, enable static port and try.

  • I already do that but no luck  >:( still getting one way voice

  • have you configured Asterisk for NAT? (sip_nat.conf i think)

Log in to reply