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amartin at xes-inc.com Guest
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Posted: Thu Apr 30, 2015 12:18 pm Post subject: [asterisk-users] OpenVPN Clients Intermittently Cannot Call |
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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
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admin at tootai.net Guest
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Posted: Thu Apr 30, 2015 4:43 pm Post subject: [asterisk-users] OpenVPN Clients Intermittently Cannot Call |
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Le 30/04/2015 19:18, Andrew Martin a écrit :
Hello
Quote: |
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
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amartin at xes-inc.com Guest
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Posted: Thu Apr 30, 2015 5:05 pm Post subject: [asterisk-users] OpenVPN Clients Intermittently Cannot Call |
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----- Original Message -----
Quote: | From: "Administrator TOOTAI" <admin@tootai.net>
To: asterisk-users@lists.digium.com
Sent: Thursday, April 30, 2015 4:43:33 PM
Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In
Quote: | 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
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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admin at tootai.net Guest
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Posted: Fri May 01, 2015 6:43 am Post subject: [asterisk-users] OpenVPN Clients Intermittently Cannot Call |
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Le 01/05/2015 00:05, Andrew Martin a écrit :
Quote: | ----- Original Message -----
Quote: | From: "Administrator TOOTAI" <admin@tootai.net>
To: asterisk-users@lists.digium.com
Sent: Thursday, April 30, 2015 4:43:33 PM
Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In
Quote: | 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
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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amartin at xes-inc.com Guest
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Posted: Mon May 04, 2015 10:00 pm Post subject: [asterisk-users] OpenVPN Clients Intermittently Cannot Call |
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----- Original Message -----
Quote: | From: "Administrator TOOTAI" <admin@tootai.net>
To: asterisk-users@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 écrit :
Quote: | ----- Original Message -----
Quote: | From: "Administrator TOOTAI" <admin@tootai.net>
To: asterisk-users@lists.digium.com
Sent: Thursday, April 30, 2015 4:43:33 PM
Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call
In
Quote: | 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@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
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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gboelter at gmail.com Guest
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Posted: Tue May 05, 2015 1:06 am Post subject: [asterisk-users] OpenVPN Clients Intermittently Cannot Call |
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
On 05/05/2015 10:59 AM, Andrew Martin wrote:
Quote: |
----- Original Message -----
Quote: | From: "Administrator TOOTAI" <admin@tootai.net> To:
asterisk-users@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 écrit :
Quote: | ----- Original Message -----
Quote: | From: "Administrator TOOTAI" <admin@tootai.net> To:
asterisk-users@lists.digium.com Sent: Thursday, April 30,
2015 4:43:33 PM Subject: Re: [asterisk-users] OpenVPN Clients
Intermittently Cannot Call In
Quote: | 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@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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--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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amartin at xes-inc.com Guest
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Posted: Tue May 05, 2015 9:46 am Post subject: [asterisk-users] OpenVPN Clients Intermittently Cannot Call |
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----- Original Message -----
Quote: | From: "Guenther Boelter" <gboelter@gmail.com>
To: asterisk-users@lists.digium.com
Sent: Tuesday, May 5, 2015 1:05:44 AM
Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In
Quote: | 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@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
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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