[asterisk-users] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?

[Dennison Williams]

On 09/22/2010 08:36 AM, Carlos Chavez wrote:
> Do you have a localnet statement in your sip.conf?  That and using
> nat=no will make sure Asterisk does not replace the IP address in the
> Invite.
>   

I just wanted to give a +1 for this response.  I am using openvpn to
connect road warriors and remote offices to a central asterisk server. 
When setting up the configuration for the road warriors I created a new
subnet for them, but forgot to include their subnet as a localnet
directive in sip.conf.  The result was that sip clients on the road
warrior network would be able to register, but then when initiating a
sip call the 200 response (to the INVITE from the client) from the
asterisk server would include a contact address for some external ip
that I did not recognize.  This hint here allowed me to fix this bug,
now calls from the road warrior subnet are coming in fine.  Thanks!