Asterisk 1.4.2 and outgoing sip providers

I am a newbie to asterisk.

I have an asterisk server that is behind a cisco pix firewall. I have opened up ports udp / tcp 5060 and rtp 10000 - 20000 ports.

When I make a phone call out through the sip provider, I can not hear the other person, but the other person can hear me. Do I need to setup a stun entry for the asterisk server since it is behind the firewall?

How can I fix this issue?

Thanks

Eric

Are you NATing? If so, you need to change your nat settings in sip.conf. I’m natting behind a CheckPoint firewall without issue with settings similar to this on a recent version of the 1.4 svn branch.

sip.conf

[general]
nat=yes ; I think you can put this here or on the sip peer of your provider
localnet=192.168.0.0/255.255.255.0 ; Your local network space
externip=4.2.2.1 ; Your external IP address

rtp.conf

[general]
rtpstart=10000
rtpend=20000

Hope this helps

and don’t forget to turn off SIP support on Cisco box!