You might also want to make sure sipx is fully updated. There were a couple of patches related to sipxbridge that were pushed out just last week.
On Mon, Feb 20, 2012 at 2:28 PM, Matt White <[email protected]>wrote: > When you say your server has a public ip address, do you mean the nic on > the server has the public ip assigned to it or that you have a firewall > that port forwards/NATS the public ip to your sipx server? If you do have > the sipx server outside of a firewall you will want to take great care > making sure iptables is done well on sipx. Otherwise you will get DoS real > fast. > > There are a couple of other settings that work in conjunction with the NAT > travesal setting. They are the server "public ip address" that is set > under the server tab and then the page and the "intranet subnets" under > the internet calling tab. > > But if your server truly has a public ip then they should be enabled. The > NAT checkbox allows sipxrelay to re-write the SIP header and inject the > "public ip" rather than the private IP. It can do this selectivity based > on the intranet subnet of the phone. It also allows it to anchor the > media. If it thinks the phones are local, it will not anchor the media > when calling between the endpoints. > > You can see if the phones show as remote by looking at the register page. > What you could do is set the public ip as the only only local subnet, and > that would make it anchor all media. > > -m > > >>> glomos-info <[email protected]> 02/20/12 12:36 PM >>> > > Dear all, > > We have deployed a Sipxecs 4.4 server (latest fixes) and are experiencing > problems with NAT traversal. > > Our server has a public IP address. > We have a SIP trunk configured to an ITSP. > We are using a combination of remote Sip phones both on NAT and without > NAT. > For supporting the NAT users, the NAT traversal option has been enabled. > > The problem is that using this configuration the non-NAT phones work OK, > but the NAT phones do not setup an RTP connection correctly (no sound both > ways). > > After we enable the 'server behind NAT' checkbox both NAT and non-NAT are > able to connect successfully. > But with the 'server behind NAT' checkbox enabled the non-NAT phones will > lose RTP connection after about 5 minutes on inbound phone calls (session > will stay, but sound drops). Outbound phone calls have no problem. > The NAT phones though work perfectly (both inbound and outbound) with the > 'server behind NAT' setting enabled. > > What are we doing wrong? What does the 'server behind NAT' checkbox > exactly do, related to NAT traversal? > Why do we have to enable it to get our NAT phones working while our server > has a public IP? > > Help is appreciated very much. > > Thanks, > GJ > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > _______________________________________________ > 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: 434.465.6833 ~~~~~~~~~~~~~~~~~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~~~~~~~~~~~~~~~~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net
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