Hey there! This is my first post in sipx-users mailing list, so I'm sorry if I do something wrong. Well.. the problem is strange enough, and I can't find out how to fix it during 2 days, hm.. So, i've got a sipX station, and it has public IP. So "Server behind NAT" is unchecked. There are about 5 users, whos phones use NAT. Everything is ok with softphones and SIP mobile phones. The only settings i fix in the softphones and SIP mobiles phones are REG server and Domain/Outbound Proxy server. Everything works fine! I use this pretty good guide _http://wiki.sipfoundry.org/display/sipXecs/Remote+User+NAT+Traversal But! The only phone (Siemens Gigaset C470 IP) doesn't work as good as i need. The scheme of my VoIP network looks like this: SIPX Server - > IP-network -> Remote NAT + Siemens Gigaset C470 IP So the sipX station does'n use any nat, but the phone does. SIP is ok, so I can call this user and he can call me. When the call goes from Siemens Gigaset C470 IP we can speak and everything is ok. But if I call him, there is no media between our phones. So the remote user can hear the ring, pick up the phone, but we can't hear each other.. I've already tried to fix everything I could. There are those types of options in the Siemens Gigaset C470 IP web GUI:
Domain Proxy server address and port REG server addres and port STUN yes/no STUN server Outbound proxy server and port I've alredy tried different combinations of those settings (even checking "use STUN"), but there is no result. No media when anyone call him. I will be happy if someone could help me to find out where the problem is. Regards, kga. _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
