In testing, I am having a hard time getting NAT Traversal to function the way I assumed it would for remote workers.
I assumed if the remote worker had a personal firewall/router it would traverse and work, but so far have not found this to be the case. In every case I can register the user and call sipx (AA, VM, other users). I can even dial out via the sip trunk. In every case dialling out via the sip trunk has broken audio (non-existent). Local users cannot call the remote worker. DID calls get sent directly to voicemail instead of ringing. The only way I have seen it work is if the remote worker phone/PC has a PUBLIC IP address and no NAT at the other end. Example: Registrations like this do not work: <sip:2...@public_ip_address_from_isp;rinstance=5734ce2f7e265247;transport=TCP;x-sipX-privcontact=PRIVATE_IP_ADDRESS_ON_LAN%3A10892%3Btransport%3DTCP> Whereas registrations like this do: <sip:2...@public_ip_address_from_isp:3062;rinstance=5fda7fd7c30a4dab;transport=TCP;x-sipX-nonat> I have an instance to want to set up two remote workers on a single Internet connection. Is this possible? _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
