I'm struggling with a SIP installation with a remote user. THe phone boots up, provisions itself from the (remote) TFTP server, performs SIP registration with the PBX, and I can make calls to the phone, it rings, they can hear me, but silence coming back.
Normally at this point I would suspect a NAT problem, but both ends of the link have Draytek Vigor routers and we've set up a site-to-site VPN between them to eliminate any NAT issues. The PBX 'sees' the device with its correct IP address, and in the PBX's dashboard it shows up as registered with no NAT.
so we have:
PBX (192.168.162.11) <--> Draytek <--> VPN <--> Draytek <--> Phone (192.168.250.9)
I would have thought that was a simple routed network. But no, apparently. I'm tearing my hair out trying to get the audio path to work correctly. As far as I can see, no NAT should mean that the traffic just flows directly from the phone to the PBX (over the VPN). I know SIP is problematic with NAT, but that shouldn't be an issue here, should it? Why isn't this working!!!??? Any thoughts would be appreciated.
The PBX is Trixbox (asterisk). The phone is Cisco 7960.