> Is the remote worker feature still a work in progress on > 3.11? I looked around a bit and didnt find anything > pertaining to it. I am having a no audio problem with it, and > Im thinking maybe i missed something in the configuration. I > tried to debug a bit as well but when trying to | > syslogviewer for some reason some of the callflow gets lost > and syslogviewer goes batty; also is there anywhere in 3.11 > to add additional syslog servers? Sorry for the numerous > questions over tha last few days but I am very excited so far > with 3.11. The conference feature is great! > > GB >
The remote NAT Traversal feature is complete and is currently undergoing verification. Having said that, the configuration pages used to manage the feature will change and as a result there isn't a whole lot of information available on the wiki that walks you through the configuration process. At the moment, the feature can be managed via sipXconfig by navigating to System->Internet Calling->NAT Traversal. Under that page, you have to configure the following: Step 1 - configuring NAT traversal feature =========================================== * "Enable NAT Traversal": checked * "Server behind NAT": check this field if your sipXecs is deployed inside a private network fronted by a NAT/Firewall. The instructions that follow assume that it is. * If the WAN-facing IP address of your NAT/Firewall if static then enter it under "Public IP Address" and blank out the "STUN server" field. If your WAN-facing IP address is dynamic (i.e. you ISP changes it from time to time), leave "Public IP Address" blank and enter a STUN server address in the "STUN Server" field (stun01.sipphone.com is a reliable public STUN server that you could use here) * Set "External Address" to the same value as "Public IP address". If "Public IP address" is blank, also blank out "External Address". * Keep all other fields as defaults. Step 2 - define your local private network topology =================================================== * This step is a bit clumsy is a good example of why we want to rework the way the feature is managed. Navigate to navigating to System->Internet Calling. There, you need to enumerate the domains and subnets that make up the private network that your sipXecs is a part of. For example, if your sipXecs has domain name sipx.example.com and is part of private network 10.10.10.0/24 then you need to have an "Intranet Domains" entry of "*.sipx.example.com" and an "Intranet subnets" entry of "10.10.10.0/24" and remove the default ones. This page has a bug whereby you cannot change that configuration unless you select a Default SBC. To get over that hurdle, you need to create a bogus unmanaged SBC but be sure *NOT* to check the "enable internet calling" Step 3 - Configure your firewall ================================ Next, you need to go on your Firewall/NAT and open pinholes and port forwarding rules that will route any incoming TCP and UDP traffic arriving on port 5060 to the IP address of your sipXecs. You also need to route traffic arriving on UDP port range 30000 to 30500 used for media to the IP address of your sipXecs. Ping me if you ruin into trouble with any of this. _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
