Depending on the sip server you are connecting to you probably need to unblock some high udp port ranges for the RTP stream to get audio.

These are usually somewhere in between 8000 and 30000. Most asterisk installs specify either 8000-9000 only or 10000-20000

Note that when you are on a connection that is not your own, you just might not be able to use SIP if lots of ports are blocked. If you need to connect phone out and about on various wifi, Skype has lots of firewall busting tricks. You can also attempt to use an STUN server with your SIP connection which might help.