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

Hello,

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.

Asterisk server LAN IP: 10.10.32.10
My internal test phone: 146 at 10.10.32.96
My external test phone: 265 at 192.168.32.10

My sip.conf for these external users is as follows:
http://pastebin.com/2b9YE7Dz


The dialplan uses this Dial() invocation when dialing either an internal or
external phone. Note that the max timeout is 12 seconds:
exten => _[12]XX,1,Dial(SIP/${EXTEN},12)


These external phones register correctly, and internal users can call these
external users, the phones ring immediately, and the call is normal. However, if
the external users try to dial an internal phone, I've observed some
different failure modes:
* operating normally: sometimes the call rings immediately, the internal user
answers, and the audio is present immediately
* ringing delay and no connection even after pickup: sometimes there's a
significant delay between when the call starts "ringing" on the
external side and when it actually starts ringing on the internal user's
phone. Consequently, the internal user only has 1 or 2 rings to answer. Even if
they do answer during this time, the line is dead and it goes to voicemail (the
next step in the dialplan)
* delay before audio is connected after answer: sometimes the internal user
answers, but there's a delay of 3-10 seconds before either party can hear
audio

I've enabled rtp and sip debug for this particular external phone
(192.168.32.10) and attached console logs from both types of these failures:
* ringing delay and no connection even after pickup:
http://pastebin.com/fe1khEmF
* delay before audio is connected after answer: http://pastebin.com/uZSMKczk

What else can I try to debug these problems? Since it is intermittent, I am not
always able to reproduce (sometimes the calls work just fine).

Thanks,

Andrew Martin

Le 30/04/2015 19:18, Andrew Martin a ?crit :
> Hello,
Hello
>
> 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

----- 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