[asterisk-users] SIP connections over OpenVPN connection get one-way voice.

[Mark Wiater]

On 4/18/2017 7:40 PM, Ernie Dunbar wrote:
> Server network: 192.168.0.0/24
> OpenVPN network: 10.8.0.0/24
> Asus network: 192.168.1.0/24
>
> The Asterisk SIP registration appears to be responding properly to 
> this - this is what I see when I do a 'sip show peer' for an Aastra 
> phone that's connecting through the VPN (Asterisk output is truncated):
>
>   ToHost       :
>   Addr->IP     : 10.8.0.6:5060

If the Asus network is 192.168.1.0/24, and the phone is registering as 
10.0.8.6, it looks like NAT is taking place. Would your asterisk server 
know how to route traffic to 192.168.1.0/24?

I've always used site-to-site OpenVPN tunnels where the vpn's terminate 
on the gateway for both the phones and the asterisk server. I've always 
had rock solid connections between phones and Asterisk.