Does your remote site with the gigaset have a router? If so, ensure any sip ALG or "helpers" are turned off within the router. Typically I would start by turning off NAT and STUN on a device, as sipx prefers to handle NAT on the server side for the UA. I am not sure if you can do this with the gigaset though.
After you have done this, you might produce a sip trace file: http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+Sipviewer and post it for more assistance. On Fri, Jul 8, 2011 at 4:48 AM, <[email protected]> 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 > 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/ > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.326.5325 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Contract Customers: http://support.myitdepartment.net <http://support.myitdepartment.net>Blog: http://blog.myitdepartment.net Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
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