Hi Ernie,
When one-way audio appear (no matters if there is a VPN or NAT server on
the diagram) I simply :
* Enable SIP debug on Asterisk server. Excecute 'sip set debug ip x.x.x.x'
on Astrisk CLI, where x.x.x.x is the IP of the phone or SIP peer you want
to debug.
* Make a test call and
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
On 2017-04-18 05:21 PM, Duncan Turnbull wrote:
Sent from my iPhone
On 19/04/2017, at 11:43 AM, Ernie Dunbar
wrote:
On 2017-04-18 03:38 PM, Duncan
Sent from my iPhone
> On 19/04/2017, at 11:43 AM, Ernie Dunbar wrote:
>
>> On 2017-04-18 03:38 PM, Duncan Turnbull wrote:
>> -- Original Message --
>> From: "Ernie Dunbar"
>> To: "'Asterisk Users Mailing List - Non-Commercial
On 2017-04-18 03:38 PM, Duncan Turnbull wrote:
-- Original Message --
From: "Ernie Dunbar"
To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
Sent:
On 2017-04-18 03:39 PM, Sebastian Nielsen wrote:
You need to ensure that traffic to the SIP box is sent to the
correct IP. Also if you use split-tunnel (eg: not redirect-gateway
def1) you must make sure NAT and traffic redirection works as is
so the
You need to ensure that traffic to the SIP box is sent to the correct IP. Also
if you use split-tunnel (eg: not redirect-gateway def1) you must make sure NAT
and traffic redirection works as is so the Asus router knows it should send the
traffic through tunnel and not via WAN.
IMPORTANT: Then
-- Original Message --
From: "Ernie Dunbar"
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: 19-Apr-17 10:25:59 AM
Subject: [asterisk-users] SIP connections over OpenVPN connection get
one-way voice.
Hi