[Help] SIP Trunk / Link configuration

Hello Everybody,

I am configuring Asterisk as my study project name Future in VoIP.

Our university has got some calling minutes from provider for this project, and Provider has only given us the IP address of there server and port 5060. I would like to use this setting and make the calls, seems I am missing some configuration in creating TRUNKS. :question:

Please help me out in configuring this. Once again I would like to brief about my setup : Its Pentium IV system with A@Home configured, I have only IP address for minute providers (no username and IP address is required) g.729 is the code. Please let me know how to configure this.

Thanks in advance