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
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 re
Thanks for the detailed info. Problem was solved by including Server B
subnet as the localnet of the Server A (OpenVPN server) and setting each
extension NAT=NO.
Your points are good guides for future problem diagnoses.
Thanks again,
Bruce
On Thu, Sep 23, 2010 at 1:56 PM, Dave Platt wrote:
>
>
> I don't think it's an endpoint issue. I think the SIP packet headers get
> over-written by the tunnel (openvpn) protocol.
I'd be rather astonished if OpenVPN itself were responsible for this.
As far as I know, OpenVPN doesn't do higher-level-protocol rewriting
of any sort. It just provides the
Calls are not going outside of the network. I had to setup up the subnet of
the other side (openvpn client) as the localnet of the Asterisk server for
Asterisk to not handle it with NAT or hand shake it with external IP.
Thanks,
-Bruce
On Wed, Sep 22, 2010 at 1:58 PM, Paul Belanger wrote:
> On
On Wed, Sep 22, 2010 at 1:46 PM, bruce bruce wrote:
> Thanks, but Carlos Chavez was right on point. This fixed the problem:
> externip=123.123.123.123
> localnet=192.168.100.0/255.255.255.0
> nat=no in each extension.
>
So now I am confused, If you have a VPN setup between sites, why are
calls goi
Thanks, but Carlos Chavez was right on point. This fixed the problem:
externip=123.123.123.123
localnet=192.168.100.0/255.255.255.0
nat=no in each extension.
Maybe combination of both or only the localnet just fixed it.
Thanks,
Bruce
On Wed, Sep 22, 2010 at 1:35 PM, Steve Edwards wrote:
> Un-
Un-top-posting...
> On Wed, Sep 22, 2010 at 1:27 AM, bruce bruce
> wrote:
> > Any feed back is appreciated.
> On Wed, Sep 22, 2010 at 9:49 AM, Paul Belanger
> wrote:
> Then configure you endpoints to use the 192.168.100.0/24 network. This
> is not an Asterisk issue, since your A
Thanks for that Carlos. I am playing with that right now. What do you
suggest localnet should say?
Server A = OpenVPN Server:
localnet=127.0.01
localnet=192.168.100.0/255.255.255.0
Where 192.168.100.0 is the DHCPd subnet of Server B (the openvpn client)
Server A doesn't have any localnet other t
I don't think it's an endpoint issue. I think the SIP packet headers get
over-written by the tunnel (openvpn) protocol.
Thanks,
Bruce
On Wed, Sep 22, 2010 at 9:49 AM, Paul Belanger wrote:
> On Wed, Sep 22, 2010 at 1:27 AM, bruce bruce wrote:
> > Any feed back is appreciated.
> >
> Then configu
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.
On Wed, 2010-09-22 at 01:27 -0400, bruce bruce wrote:
> Hi Everyone,
>
>
> I have setup an OpenVPN tunnel between Server A (running Asterisk) and
> Ser
On Wed, Sep 22, 2010 at 1:27 AM, bruce bruce wrote:
> Any feed back is appreciated.
>
Then configure you endpoints to use the 192.168.100.0/24 network.
This is not an Asterisk issue, since your Aastra 55i/2.5.2.1500 is
sending the INVITE message.
--
Paul Belanger | dCAP
Polybeacon | Consultant
J
Thanks for the feedback. I thought about that but it's not an option for me
right now.
Any other ways folks?
Thanks
On Wed, Sep 22, 2010 at 4:06 AM, Roger Burton West wrote:
> On Wed, Sep 22, 2010 at 01:27:07AM -0400, bruce bruce wrote:
> >I have setup an OpenVPN tunnel between Server A (runnin
On Wed, Sep 22, 2010 at 01:27:07AM -0400, bruce bruce wrote:
>I have setup an OpenVPN tunnel between Server A (running Asterisk) and
>Server B suppling it's SIP Phones with DHCP pool of IPs.
Have you considered running Asterisk on Server B as well, and using IAX
to trunk between them? This is work
Hi Everyone,
I have setup an OpenVPN tunnel between Server A (running Asterisk) and
Server B suppling it's SIP Phones with DHCP pool of IPs.
So, the tunnel is established nicely and everyone can ping others. "sip show
peers" shows the local subnet of the SIP Phones registered (192.168.100.0/24
).
15 matches
Mail list logo
