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

[Ming-Wei Shih]

Michael Wang wrote:

>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.
>  
>
try this in your sip.conf

disallow=all
allow=ulaw
allow=alaw
nat=yes

or use a STUN server

Ming-Wei