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Andrew Martin

2015-May-05 02:59 UTC

[asterisk-users] OpenVPN Clients Intermittently Cannot Call In

----- 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

>

Guenther Boelter

2015-May-05 06:05 UTC

head link

[asterisk-users] OpenVPN Clients Intermittently Cannot Call In

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

On 05/05/2015 10:59 AM, Andrew Martin wrote:> 
> 
> ----- 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
Hi Andrew,

is this a linux machine? If so, check your NIC with ifconfig for
hardware errors.

Guenther


- -- 
DavaoSOFT, the home of ERPel
ERPel, das deutsche Warenwirtschaftssystem fuer LINUX
http://www.davaosoft.com
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Andrew Martin

2015-May-05 14:46 UTC

head link

[asterisk-users] OpenVPN Clients Intermittently Cannot Call In

----- Original Message -----> From: "Guenther Boelter" <gboelter at gmail.com>
> To: asterisk-users at lists.digium.com
> Sent: Tuesday, May 5, 2015 1:05:44 AM
> Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In
> 
> > 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
> 
> Hi Andrew,
> 
> is this a linux machine? If so, check your NIC with ifconfig for
> hardware errors.
> 
> Guenther
> 
Guenther,

Yes, this machine is running CentOS 6.4 (see my original post for more
details). This asterisk server has 2x gigabit NICs set up in a bond with
bond mode 1.

Both ifconfig and ethtool do not report any hardware errors,
although they do show a few checksum errors:
eth0      Link encap:Ethernet  HWaddr 00:11:22:33:44:55
          UP BROADCAST RUNNING SLAVE MULTICAST  MTU:1500  Metric:1
          RX packets:467927100 errors:0 dropped:0 overruns:1 frame:0
          TX packets:304724661 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:1000
          RX bytes:131747094082 (122.6 GiB)  TX bytes:93869585242 (87.4 GiB)
          Memory:fb920000-fb940000

eth1      Link encap:Ethernet  HWaddr AA:BB:CC:DD:EE:FF
          UP BROADCAST RUNNING SLAVE MULTICAST  MTU:1500  Metric:1
          RX packets:41250363 errors:0 dropped:0 overruns:0 frame:0
          TX packets:3467 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:1000
          RX bytes:5190889937 (4.8 GiB)  TX bytes:1594075 (1.5 MiB)
          Memory:fb900000-fb920000
>From ethtool -S eth0:     tx_smbus: 164709
     rx_smbus: 119082408
     dropped_smbus: 104036

     rx_queue_0_packets: 97532982
     rx_queue_0_bytes: 16800645524
     rx_queue_0_drops: 1
     rx_queue_0_csum_err: 0
     rx_queue_0_alloc_failed: 0

     rx_queue_7_packets: 53850556
     rx_queue_7_bytes: 12797600155
     rx_queue_7_drops: 0
     rx_queue_7_csum_err: 41
     rx_queue_7_alloc_failed: 0

Thanks,

Andrew

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

[asterisk-users] OpenVPN Clients Intermittently Cannot Call In

[asterisk-users] OpenVPN Clients Intermittently Cannot Call In

[asterisk-users] OpenVPN Clients Intermittently Cannot Call In

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