I tried everything i could, and it still doesn't work. An interesting thing is that there are 2 more connections via SIP between this phone and other SIP-providers, and everything is fine! >> >> On Fri, Jul 8, 2011 at 4:48 AM, wrote: >> 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 >> [3] >> 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] [4] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ [5] >> >> -- >> ====================== >> Tony Graziano, Manager >> Telephone: 434.984.8430 >> sip: [email protected] [6] >> Fax: 434.326.5325 >> >> Email: [email protected] [7] >> >> LAN/Telephony/Security and Control Systems Helpdesk: >> Telephone: 434.984.8426 >> sip: [email protected] [8] >> >> Helpdesk Contract Customers: >> http://support.myitdepartment.net [9] >> >> [10]Blog: >> http://blog.myitdepartment.net [11] >> >> Linked-In >> Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 [12] >> >> >> Links: >> ------ >> [1] >> >> http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+Sipviewer >> [2] mailto:[email protected] >> [3] >> http://wiki.sipfoundry.org/display/sipXecs/Remote+User+NAT+Traversal >> [4] mailto:[email protected] >> [5] http://list.sipfoundry.org/archive/sipx-users/ >> [6] mailto:[email protected] >> [7] mailto:[email protected] >> [8] mailto:[email protected] >> [9] http://support.myitdepartment.net >> [10] http://support.myitdepartment.net >> [11] http://blog.myitdepartment.net >> [12] http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
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