[asterisk-users] asterisk and openvpn and sip

[Darrick Hartman (lists)]

Do you have 'canreinvite=no' in your sip.conf entry for this phone?  If 
not, you should.

On 06/18/2009 07:55 AM, Giorgio Incantalupo wrote:
> Hi John,
>
> I already have the ccd dir with the iroute (mandatory for routing to
> pc/phone connected to vpn client). During the last test I could register
> and  make a call but voice disappears after 1, 2 seconds. I'm trying to
> understand if it is a bandwidth problem. At the moment I have my phone
> connected to the openvpn client (which is its gateway) but I have to use
> the vpn ip (10.0.0.1) to register the phone, the openvpn server local ip
> (192.168.1.12) is not working. I suppose it is a  sip protocol problem:
> I had to change the sip.conf setting nat=yes to make the phone dial and
> domain = 10.0.0.1 to make the voice pass (or at least the first 2 seconds).
> I keep on working on the vpn since it seems so little is missing to have
> a clear conversation. Let me know if your tests are successfull.

-- 
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com