[Asterisk-Users] question regarding Asterisk. X-Lite, and firewall

[Michael Wang]

Hello,

I have a one-way audio problem. If any one can give me a clue on how to
solve it, I'd highly appreciate.

My configuration is:

Both Asterisk server and a SIP phone run within a LAN. Asterisk:
CVS-HEAD-06/27/04-11:42:23. SIP phone is X-Lite release 1103m build stamp
14262. The Linux box that running Asterisk server is RedHat 2.4.18-14.

Asterisk server runs on IP: 192.168.1.102. X-Lite (phone A) is on Win2K,
with IP 192.168.1.100. They are both behind a router with dynamic IP
address. Assume its public IP is aaa.bbb.ccc.ddd.

I have another X_Lite SIP phone (phone B) that is NOT in the LAN I mentioned
above. Rather, it has its own public IP address, say eee.fff.ggg.hhh.

I have configured the router to forward all traffic to its port 5161 to
Asterisk server's 5060 port, and configured SIP phone A to use
192.168.1.102:5060 and phone B aaa.bbb.ccc.ddd:5161 as proxy server
respectively. Both phones registered successfully.

Now, I used phone B to call phone A. The entire SIP hand-shake went through
successfully. However, I can only get voice from phone A to phone B, not the
other direction. I found that RTP traffic went from phone A -> Asterisk ->
phone B. However, on the other direction, phone B tried to use 192.168.1.102
as destination of Asterisk to send voice too. Obviously, the IP is a private
IP, hence, is not reachable.

How do I change configuration of Asterisk so that phone B can use
aaa.bbb.ccc.ddd as RTP destination, instead of the private IP address?

By the way, both directions use UDP protocol.

Thanks!
        Michael Wang
        mwang at empirestar.com
          2004-07-20