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

[bruce bruce]

Glad to hear it helped you Dennison.

VPN is such a confusing beast to lots of people I think and hence the
responses to this thread were all sort of work around and sometimes
off-topic. It's also not well documented or maybe the feature is not widely
used within the Asterisk community. I think it would be very good if some
standard guidelines become available from the Asterisk side on this.

Good day,
Bruce


On Wed, Oct 6, 2010 at 7:33 PM, Dennison Williams <
dennison.williams at gmail.com> wrote:

> 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!
>
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