Recently I have installed and configured asterisk server.
I am able to make incoming and outgoing calls from local network i.e. intranet.
When i am making calls from outside network, call is getting established, but unable to hear the voice from anyside.
I am having firewall. In that I have forwared the port 5060 ( UDP Port) , also forwarded teh port 10000-20000 ( UDP Ports ) for RTP which is required for audio transmission.
Complicated problem. If you mean device behind firewall - then you can set on that device STUN.
You can watch packets using something like tcpdump or iptraf. And see what is wrong. In worst cases I used iptraf on Asterisk server and on router behind which was my device.
on some of my clients if they are behind a NAT (since the Asteisk server is also behind a NAT) and that has taken care of it. USUALLY an established call, but no voice is a perfect example of the limitations of the SIP UDP protocal and the need for a STUN server.
ALSO if your clients are behind a NAT make sure that the NAT=YES is in their sip.conf setting for that extension.