because sip provider has a way handle your device randomizing portal. that's what a sbc does.. that device was designed for consumer use . sipx is a system geared toward businesses. either get border controller with that phone or get another phone. On Jul 12, 2011 3:40 AM, <[email protected]> wrote: > 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 > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/
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