[asterisk-gui] asterisk 1.4.2 and outgoing sip providers

[Eric Swanson]

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
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