Hi Tony, We did already update with the last fixes. No success.
Thanks, GJ Van: [email protected] [mailto:[email protected]] Namens Tony Graziano Verzonden: maandag 20 februari 2012 20:35 Aan: Discussion list for users of sipXecs software Onderwerp: Re: [sipx-users] Welcome to the "sipx-users" mailing list 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]<mailto:[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]<mailto:[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]<mailto:[email protected]> List Archive: http://list.sipfoundry.org/archive/sipx-users/ _______________________________________________ sipx-users mailing list [email protected]<mailto:[email protected]> List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~~~~~~~~~~~~~~~~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected]<mailto:[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]<mailto:[email protected]> Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net
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