Michael Wang
2004-Jul-20 10:31 UTC
[Asterisk-Users] question regarding Asterisk. X-Lite, and firewall
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@empirestar.com ??????????2004-07-20
Ming-Wei Shih
2004-Jul-20 12:35 UTC
[Asterisk-Users] question regarding Asterisk. X-Lite, and firewall
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
Michael Wang
2004-Jul-21 08:14 UTC
[Asterisk-Users] question regarding Asterisk. X-Lite, and firewall
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@empirestar.com
2004-07-20