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 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users