asterisk users - May 2015 - OpenVPN Clients Intermittently Cannot Call In

----- Original Message -----
> From: "Administrator TOOTAI" <admin at tootai.net>
> To: asterisk-users at lists.digium.com
> Sent: Thursday, April 30, 2015 4:43:33 PM
> Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In
> 
> > I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and
> > internal phones are located on the 10.10.32.0/21 LAN subnet. I have
many
> > internal SIP phones, which appear to be working correctly. I have a
few
> > external phones (Yealink SIP-T32G or other Yealink model) on
> > 192.168.32.0/24 which have an OpenVPN client configured on them that
> > connects back to the LAN network through a pfSense gateway with
OpenVPN
> > configured on it.
> 
> I faced problems with pfsense -no VPN involved- and finally installed
> siproxd on it. Also set the firewall mode to conservative.
Daniel,

Thanks for the information. Do you have an example or documentation on the
siproxd configuration that you used?

Thanks,

Andrew

Le 01/05/2015 00:05, Andrew Martin a ?crit :
> ----- Original Message -----
>> From: "Administrator TOOTAI" <admin at tootai.net>
>> To: asterisk-users at lists.digium.com
>> Sent: Thursday, April 30, 2015 4:43:33 PM
>> Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot
Call In
>>
>>> I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server
and
>>> internal phones are located on the 10.10.32.0/21 LAN subnet. I have
many
>>> internal SIP phones, which appear to be working correctly. I have a
few
>>> external phones (Yealink SIP-T32G or other Yealink model) on
>>> 192.168.32.0/24 which have an OpenVPN client configured on them
that
>>> connects back to the LAN network through a pfSense gateway with
OpenVPN
>>> configured on it.
>>
>> I faced problems with pfsense -no VPN involved- and finally installed
>> siproxd on it. Also set the firewall mode to conservative.
>
> Daniel,
>
> Thanks for the information. Do you have an example or documentation on the
> siproxd configuration that you used?
No, just follow the basis of the parameters given by the package. If I 
remember, SIP use the proxy siproxd and RTP is direct.

Another solution I used on an not stable xDSL line, was to install 
asterisk on pfsense, this asterisk taking only care on the local traffic 
(call from local extension to local extension). The asterisk register 
with the main one as a trunk for incoming/outgoing calls. Worked too.

-- 
Daniel

----- Original Message -----
> From: "Administrator TOOTAI" <admin at tootai.net>
> To: asterisk-users at lists.digium.com
> Sent: Friday, May 1, 2015 6:42:38 AM
> Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In
> 
> Le 01/05/2015 00:05, Andrew Martin a ?crit :
> > ----- Original Message -----
> >> From: "Administrator TOOTAI" <admin at tootai.net>
> >> To: asterisk-users at lists.digium.com
> >> Sent: Thursday, April 30, 2015 4:43:33 PM
> >> Subject: Re: [asterisk-users] OpenVPN Clients Intermittently
Cannot Call
> >> In
> >>
> >>> I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk
server and
> >>> internal phones are located on the 10.10.32.0/21 LAN subnet. I
have many
> >>> internal SIP phones, which appear to be working correctly. I
have a few
> >>> external phones (Yealink SIP-T32G or other Yealink model) on
> >>> 192.168.32.0/24 which have an OpenVPN client configured on
them that
> >>> connects back to the LAN network through a pfSense gateway
with OpenVPN
> >>> configured on it.
> >>
> >> I faced problems with pfsense -no VPN involved- and finally
installed
> >> siproxd on it. Also set the firewall mode to conservative.
> >
> > Daniel,
> >
> > Thanks for the information. Do you have an example or documentation on
the
> > siproxd configuration that you used?
> 
> No, just follow the basis of the parameters given by the package. If I
> remember, SIP use the proxy siproxd and RTP is direct.
> 
Looking into it further, in my case it does not appear to be a NATing issue,
since running OpenVPN from pfSense means there's no NATing occurring between
the clients or between the clients and the asterisk server.

Although I was unable to reproduce the problems, I did notice some packet loss
and jitter in "sip show channelstats", here is a sample:
Peer             Call ID      Duration Recv: Pack  Lost       (     %) Jitter
Send: Pack  Lost       (     %) Jitter
192.168.32.26    446613544 at 1  00:03:03 0000000094  0000004238 (97.83%) 0.0000
0000000000  0000000244 ( 0.00%) 0.0000
192.168.32.38    5b2ebdc92fd  00:03:03 0000000059  0000000001 ( 1.67%) 0.0000
0000000000  0000000091 ( 0.00%) 0.0028

I was unable to find documentation each of these columns, but the high
percentage
of loss for received packets for 192.168.32.26 seems suspicious. Do these
statistics
indicate a problem?

Thanks,

Andrew

>