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dcunningham at voisoni...
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 Posted: Tue Jan 21, 2014 6:23 am Post subject: [asterisk-users] Asterisk not receiving call from VPN addres |
 |
|
Hi Duncan,
We have "sip set debug on" and nothing is shown, even though tcpdump/ngrep on the same server does. It's very strange.
The output of "ip address list" is:
[root]# ip address list
1: lo: <LOOPBACK,UP,LOWER_UP> mtu 16436 qdisc noqueue state UNKNOWN
link/loopback 00:00:00:00:00:00 brd 00:00:00:00:00:00
inet 127.0.0.1/8 scope host lo
inet6 ::1/128 scope host
valid_lft forever preferred_lft forever
2: eth0: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500 qdisc mq state UP qlen 1000
link/ether 44:1e:a1:4e:2f:b8 brd ff:ff:ff:ff:ff:ff
inet 103.y.y.19/24 brd 103.y.y.255 scope global eth0
inet6 fe80::461e:a1ff:fe4e:2fb8/64 scope link
valid_lft forever preferred_lft forever
3: eth1: <BROADCAST,MULTICAST> mtu 1500 qdisc noop state DOWN qlen 1000
link/ether 44:1e:a1:4e:2f:ba brd ff:ff:ff:ff:ff:ff
4: eth2: <BROADCAST,MULTICAST> mtu 1500 qdisc noop state DOWN qlen 1000
link/ether 44:1e:a1:4f:30:a4 brd ff:ff:ff:ff:ff:ff
5: eth3: <BROADCAST,MULTICAST> mtu 1500 qdisc noop state DOWN qlen 1000
link/ether 44:1e:a1:4f:30:a6 brd ff:ff:ff:ff:ff:ff
6: tun0: <POINTOPOINT,MULTICAST,NOARP,UP,LOWER_UP> mtu 1500 qdisc pfifo_fast state UNKNOWN qlen 100
link/[65534]
inet 172.x.x.14 peer 172.x.x.13/32 scope global tun0
The output of "netstat -rn" is:
[root]# netstat -rn
Kernel IP routing table
Destination Gateway Genmask Flags MSS Window irtt Iface
172.x.x.10 172.x.x.13 255.255.255.255 UGH 0 0 0 tun0
172.x.x.13 0.0.0.0 255.255.255.255 UH 0 0 0 tun0
172.x.x.1 172.x.x.13 255.255.255.255 UGH 0 0 0 tun0
172.x.x.18 172.x.x.13 255.255.255.255 UGH 0 0 0 tun0
192.168.234.0 172.x.x.13 255.255.255.0 UG 0 0 0 tun0
192.168.235.0 172.x.x.13 255.255.255.0 UG 0 0 0 tun0
103.y.y.0 0.0.0.0 255.255.255.0 U 0 0 0 eth0
169.z.z.0 0.0.0.0 255.255.0.0 U 0 0 0 eth0
172.21.0.0 172.x.x.13 255.255.0.0 UG 0 0 0 tun0
10.0.0.0 172.x.x.13 255.0.0.0 UG 0 0 0 tun0
0.0.0.0 103.y.y.1 0.0.0.0 UG 0 0 0 eth0
On 21 January 2014 17:44, Duncan Turnbull <duncan@e-simple.co.nz (duncan@e-simple.co.nz)> wrote:
| Quote: | Cool
That looks like it is arriving at Asterisk - are you sure asterisk is not getting it? If you turn on sip debug in asterisk can you see the SIP packets? It maybe asterisk is ignoring them or replying to them but its going out an interface you hadn’t thought of, I have had that a few times.
I should have mentioned to print out your route table and ifconfig. Asterisk can reply on a different address to the original destination especially if it came through a tunnel. Often it will be the tunnel interface address. Usually then we set the secondary address as the outbound proxy on the phone so the phone will also respond to it.
Cheers Duncan
On 21/01/2014, at 7:18 pm, David Cunningham <dcunningham@voisonics.com (dcunningham@voisonics.com)> wrote:
| Quote: | Hi Duncan,
Thank you for your reply. Here's the netstat:
[root]# netstat -udpln | grep asterisk
udp 0 0 0.0.0.0:5000 0.0.0.0:* 6672/asterisk
udp 0 0 0.0.0.0:4520 0.0.0.0:* 6672/asterisk
udp 0 0 0.0.0.0:5060 0.0.0.0:* 6672/asterisk
udp 0 0 0.0.0.0:4569 0.0.0.0:* 6672/asterisk
Here's the tcpdump (tcpdump udp port 5060 -A -n -nn -i tun0) from the Kamailio server:
17:13:17.103771 IP 103.x.x.x.5060 > 172.y.y.y.5060: SIP, length: 1228
E.......@.>/g.v.............INVITE sip:*1@172.y.y.y:5060;transport=udp SIP/2.0
Record-Route: <sip:103.x.x.x;lr=on>
Via: SIP/2.0/UDP 103.x.x.x;branch=z9hG4bK584f.d0387c07.0
Via: SIP/2.0/UDP 192.168.1.40:5060;received=203.z.z.z;rport=5060;branch=z9hG4bK274588850
From: <sip:9067273@103.x.x.x>;tag=1880695235
To: <sip:*1@103.x.x.x>
Call-ID: 1898224288
Here's the tcpdump (tcpdump udp port 5060 -A -n -nn -i tun0) from the Asterisk server:
17:13:17.093676 IP 103.x.x.x.5060 > 172.y.y.y.5060: SIP, length: 1228
E.......?.?/g.v.............INVITE sip:*1@172.y.y.y:5060;transport=udp SIP/2.0
Record-Route: <sip:103.x.x.x;lr=on>
Via: SIP/2.0/UDP 103.x.x.x;branch=z9hG4bK584f.d0387c07.0
Via: SIP/2.0/UDP 192.168.1.40:5060;received=203.z.z.z;rport=5060;branch=z9hG4bK274588850
From: <sip:9067273@103.x.x.x>;tag=1880695235
To: <sip:*1@103.x.x.x>
Call-ID: 1898224288
|
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dcunningham at voisoni...
Guest
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| Posted: Tue Jan 21, 2014 6:24 am Post subject: [asterisk-users] Asterisk not receiving call from VPN addres |
|
|
Hi Larry,
No, they are on separate machines.
On 21 January 2014 17:54, Larry Moore <lmoore@omninet.net.au (lmoore@omninet.net.au)> wrote:
| Quote: | Is Kamalio running on the same system as Asterisk?
On 21/01/2014 2:41 PM, David Cunningham wrote:
| Quote: | Hi Larry,
Thanks for the reply. We have all of those settings left out of our
sip.conf, so this should allow everything, right?
On 21 January 2014 17:38, Larry Moore <lmoore@omninet.net.au (lmoore@omninet.net.au)
<mailto:lmoore@omninet.net.au (lmoore@omninet.net.au)>> wrote:
Have you checked your localnet=, deny=, permit=, contactdeny= &
contactpermit= settings?
My 2c worth.
On 20/01/2014 10:51 AM, David Cunningham wrote:
Hi,
We have a Kamailio and Asterisk cluster, both machines being on
a real
103.x IP address and also on a 172.x OpenVPN address.
The problem is that when Kamailo receives a call from the VPN and
forwards it to the Asterisk server on it's 103.x address,
Asterisk never
sees the call.
If Kamailio receives a call from the VPN and forwards the call
to the
Asterisk server on it's 172.x address then it works. However, if the
call isn't from the VPN then forwarding it to the 172.x address
doesn't
work. So basically the problem is going between the real network
and the
VPN.
The question is, how can we make this work when calls are
received on
either network on the Kamailio server and are forwarded to Asterisk?
Using ngrep on the Asterisk server we see that it does receive the
INVITE, but Asterisk's logging shows no sign it at all. We guess
it's a
Linux networking issue rather than Asterisk's fault, but don't know
where to fix it. We do have net.ipv4.ip_forward = 1 on both the
Kamailio
and Asterisk servers.
Thanks in advance for any help.
The ngrep on the Asterisk server:
U 2014/01/17 13:15:[url=tel:15.599557%20172]15.599557 172[/url]
<tel:15.599557%20172>.x.x.x:5060 -> 103.y.y.y:5060
INVITE sip:9067268@103.y.y.y:5060;__transport=udp SIP/2.0.
Record-Route: <sip:172.x.x.x;lr=on>.
Via: SIP/2.0/UDP 172.x.x.x;branch=z9hG4bK50c7.__f49ceb73.0.
Via: SIP/2.0/UDP
192.z.z.z:5062;rport=5062;__branch=z9hG4bK806710997.
From: "9067271" <sip:9067271@172.x.x.x>;tag=__198791249.
To: <sip:9067268@172.x.x.x>.
Call-ID: 1905625787@192.z.z.z.
...
172.x.x.x is the Kamailio server's VPN address
103.y.y.y is the Asterisk server's real address
192.z.z.z is the calling phone's LAN address
--
David Cunningham, Voisonics
http://voisonics.com/
USA: [url=tel:%2B1%20213%20221%201092]+1 213 221 1092[/url] <tel:%2B1%20213%20221%201092>
UK: [url=tel:%2B44%20%280%29%2020%203298%201642]+44 (0) 20 3298 1642[/url] <tel:%2B44%20%280%29%2020%203298%201642>
Australia: [url=tel:%2B61%20%280%29%202%208063%209019]+61 (0) 2 8063 9019[/url]
<tel:%2B61%20%280%29%202%208063%209019>
--
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USA: [url=tel:%2B1%20213%20221%201092]+1 213 221 1092[/url]
UK: [url=tel:%2B44%20%280%29%2020%203298%201642]+44 (0) 20 3298 1642[/url]
Australia: [url=tel:%2B61%20%280%29%202%208063%209019]+61 (0) 2 8063 9019[/url]
|
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USA: +1 213 221 1092
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Australia: +61 (0) 2 8063 9019 |
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andres at telesip.net
Guest
|
| Posted: Tue Jan 21, 2014 9:40 am Post subject: [asterisk-users] Asterisk not receiving call from VPN addres |
|
|
David,
It seems to me that Asterisk is not seeing/binding to your VPN
interface. You need to debug that first. I would set en explicit bind
statement in sip.conf to the VPN interface address and nothing else.
Then start your asterisk and watch the log messages. It should confirm
that it cannot bind to that address. If it does bind, then try your test
again and asterisk should see the SIP packets coming in.
--
Technical Support
http://www.cellroute.net
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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paul.belanger at polyb...
Guest
|
| Posted: Tue Jan 21, 2014 10:47 am Post subject: [asterisk-users] Asterisk not receiving call from VPN addres |
|
|
On Tue, Jan 21, 2014 at 12:40 AM, David Cunningham
<dcunningham@voisonics.com> wrote:
| Quote: | Hi Paul,
Using ngrep/tcpdump shows the packet clearly going from the Kamailio server
and arriving at the Asterisk server. This is why it's a mystery that
Asterisk doesn't see the call coming in. We tried removing the firewall (so
iptables -L shows no rules at all) but that didn't help unfortunately.
|
You might think Kamailio is transmitting it to Asterisk, however
without looking at the actually routing tables on Kamailio you'll
never know if it actually made it to Asterisk. Again, we need a pcap
trace on both Kamailio and Asterisk, plus what your routes look like
(route -n), for a call. It will show us clearly what is happening.
This all sounds like a routing issue, so your network admins should be
able to help troubleshoot.
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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paul.belanger at polyb...
Guest
|
| Posted: Tue Jan 21, 2014 10:55 am Post subject: [asterisk-users] Asterisk not receiving call from VPN addres |
|
|
On Tue, Jan 21, 2014 at 10:47 AM, Paul Belanger
<paul.belanger@polybeacon.com> wrote:
| Quote: | On Tue, Jan 21, 2014 at 12:40 AM, David Cunningham
<dcunningham@voisonics.com> wrote:
| Quote: | Hi Paul,
Using ngrep/tcpdump shows the packet clearly going from the Kamailio server
and arriving at the Asterisk server. This is why it's a mystery that
Asterisk doesn't see the call coming in. We tried removing the firewall (so
iptables -L shows no rules at all) but that didn't help unfortunately.
|
You might think Kamailio is transmitting it to Asterisk, however
without looking at the actually routing tables on Kamailio you'll
never know if it actually made it to Asterisk. Again, we need a pcap
trace on both Kamailio and Asterisk, plus what your routes look like
(route -n), for a call. It will show us clearly what is happening.
This all sounds like a routing issue, so your network admins should be
able to help troubleshoot.
|
I finally re-read the complete thread. When are you starting the VPN
on your Asterisk server, before or after Asterisk has started? If
after, and you are binding to 0.0.0.0, it is likely Asterisk is not
actually bound to your tun0 interface. So, for a test, explicitly
have asterisk listen only on the tun0 interface, retry your call.
Or setup your tunnel, then stop Asterisk and start it again, that
should cause it to bind properly.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belanger@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger
--
_____________________________________________________________________
-- 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
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dcunningham at voisoni...
Guest
|
| Posted: Tue Jan 21, 2014 4:39 pm Post subject: [asterisk-users] Asterisk not receiving call from VPN addres |
|
|
Hi Andres,
Thanks for the idea. We did send bindaddr to the VPN address and restarted Asterisk, but unfortunately that didn't solve the issue. Asterisk didn't complain, but still the "sip set debug on" didn't show the packets.
On 22 January 2014 01:40, Andres <andres@telesip.net (andres@telesip.net)> wrote:
| Quote: | David,
It seems to me that Asterisk is not seeing/binding to your VPN interface. You need to debug that first. I would set en explicit bind statement in sip.conf to the VPN interface address and nothing else. Then start your asterisk and watch the log messages. It should confirm that it cannot bind to that address. If it does bind, then try your test again and asterisk should see the SIP packets coming in.
--
Technical Support
http://www.cellroute.net
--
_____________________________________________________________________
-- 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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|
--
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019 |
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dcunningham at voisoni...
Guest
|
| Posted: Tue Jan 21, 2014 4:39 pm Post subject: [asterisk-users] Asterisk not receiving call from VPN addres |
|
|
Hi Paul,
Thanks, we did try restarting Asterisk after the VPN was up but that didn't solve the issue either.
On 22 January 2014 02:55, Paul Belanger <paul.belanger@polybeacon.com (paul.belanger@polybeacon.com)> wrote:
| Quote: | On Tue, Jan 21, 2014 at 10:47 AM, Paul Belanger
<paul.belanger@polybeacon.com (paul.belanger@polybeacon.com)> wrote:
| Quote: | On Tue, Jan 21, 2014 at 12:40 AM, David Cunningham
<dcunningham@voisonics.com (dcunningham@voisonics.com)> wrote:
| Quote: | Hi Paul,
Using ngrep/tcpdump shows the packet clearly going from the Kamailio server
and arriving at the Asterisk server. This is why it's a mystery that
Asterisk doesn't see the call coming in. We tried removing the firewall (so
iptables -L shows no rules at all) but that didn't help unfortunately.
|
You might think Kamailio is transmitting it to Asterisk, however
without looking at the actually routing tables on Kamailio you'll
never know if it actually made it to Asterisk. Again, we need a pcap
trace on both Kamailio and Asterisk, plus what your routes look like
(route -n), for a call. It will show us clearly what is happening.
This all sounds like a routing issue, so your network admins should be
able to help troubleshoot.
|
I finally re-read the complete thread. When are you starting the VPN
on your Asterisk server, before or after Asterisk has started? If
after, and you are binding to 0.0.0.0, it is likely Asterisk is not
actually bound to your tun0 interface. So, for a test, explicitly
have asterisk listen only on the tun0 interface, retry your call.
Or setup your tunnel, then stop Asterisk and start it again, that
should cause it to bind properly.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belanger@polybeacon.com (paul.belanger@polybeacon.com) | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger
--
_____________________________________________________________________
-- 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
|
--
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019 |
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asterisk.org at sedwar...
Guest
|
| Posted: Tue Jan 21, 2014 5:11 pm Post subject: [asterisk-users] Asterisk not receiving call from VPN addres |
|
|
(Please don't top-post.)
On Wed, 22 Jan 2014, David Cunningham wrote:
| Quote: | We did send bindaddr to the VPN address and restarted Asterisk, but
unfortunately that didn't solve the issue. Asterisk didn't complain, but
still the "sip set debug on" didn't show the packets.
|
Have you confirmed via 'netstat' (or some other system level toop) that
Asterisk is actually listening to UDP port 5060 on the VPN IP address?
--
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards sedwards@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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dcunningham at voisoni...
Guest
|
| Posted: Tue Jan 21, 2014 5:18 pm Post subject: [asterisk-users] Asterisk not receiving call from VPN addres |
|
|
On 22 January 2014 09:11, Steve Edwards <asterisk.org@sedwards.com (asterisk.org@sedwards.com)> wrote:
| Quote: | (Please don't top-post.)
On Wed, 22 Jan 2014, David Cunningham wrote:
| Quote: | We did send bindaddr to the VPN address and restarted Asterisk, but unfortunately that didn't solve the issue. Asterisk didn't complain, but still the "sip set debug on" didn't show the packets.
|
Have you confirmed via 'netstat' (or some other system level toop) that Asterisk is actually listening to UDP port 5060 on the VPN IP address?
--
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards sedwards@sedwards.com (sedwards@sedwards.com) Voice: [url=tel:%2B1-760-468-3867]+1-760-468-3867[/url] PST
Newline Fax: [url=tel:%2B1-760-731-3000]+1-760-731-3000[/url]
--
_____________________________________________________________________
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|
Hi Steve,
When we have bindport = 172.x.x.14 then "netstat -udpln" shows the following. When bindport is 0.0.0.0 then netstat shows it listening on 0.0.0.0 as you'd expect.
udp 0 0 172.x.x.14:5060 0.0.0.0:* 18114/asterisk
--
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019 |
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paul.belanger at polyb...
Guest
|
| Posted: Tue Jan 21, 2014 6:31 pm Post subject: [asterisk-users] Asterisk not receiving call from VPN addres |
|
|
On Tue, Jan 21, 2014 at 5:18 PM, David Cunningham
<dcunningham@voisonics.com> wrote:
| Quote: | On 22 January 2014 09:11, Steve Edwards <asterisk.org@sedwards.com> wrote:
| Quote: |
(Please don't top-post.)
On Wed, 22 Jan 2014, David Cunningham wrote:
| Quote: | We did send bindaddr to the VPN address and restarted Asterisk, but
unfortunately that didn't solve the issue. Asterisk didn't complain, but
still the "sip set debug on" didn't show the packets.
|
Have you confirmed via 'netstat' (or some other system level toop) that
Asterisk is actually listening to UDP port 5060 on the VPN IP address?
--
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards sedwards@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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|
Hi Steve,
When we have bindport = 172.x.x.14 then "netstat -udpln" shows the
following. When bindport is 0.0.0.0 then netstat shows it listening on
0.0.0.0 as you'd expect.
udp 0 0 172.x.x.14:5060 0.0.0.0:*
18114/asterisk
--
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019
|
At this point in time, you'll need to show us a .pcap on the Asterisk
box, when you make a call to it via Kamailio.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belanger@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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dcunningham at voisoni...
Guest
|
| Posted: Tue Jan 21, 2014 7:01 pm Post subject: [asterisk-users] Asterisk not receiving call from VPN addres |
|
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Hi Paul,
Thanks for the reply. What are you looking for in the PCAP, that isn't in the tcpdump earlier in the thread? I just want to make sure we gather the information required.
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andres at telesip.net
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| Posted: Tue Jan 21, 2014 7:34 pm Post subject: [asterisk-users] Asterisk not receiving call from VPN addres |
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On 1/21/14, 4:38 PM, David Cunningham wrote:
| Quote: | Hi Andres,
Thanks for the idea. We did send bindaddr to the VPN address and restarted Asterisk, but unfortunately that didn't solve the issue. Asterisk didn't complain, but still the "sip set debug on" didn't show the packets.
|
Ok, that is progress though. At this point we know that the OS is receiving the packet and Asterisk is listening on that interface and port. I know you already removed the firewall so that would not be the issue. My next guess is Asterisk is looking at the packet and dropping it because it believes it is not meant for it (Kamalio config issue), so try a simple test to confirm this. Just configure a remote IP phone/softphone via that same VPN interface to simulate a remote SIP endpoint. If the SIP phone works fine but Kamalio does not, that will clearly tell you where the problem lies.
| Quote: |
On 22 January 2014 01:40, Andres <andres@telesip.net (andres@telesip.net)> wrote:
| Quote: | David,
It seems to me that Asterisk is not seeing/binding to your VPN interface. You need to debug that first. I would set en explicit bind statement in sip.conf to the VPN interface address and nothing else. Then start your asterisk and watch the log messages. It should confirm that it cannot bind to that address. If it does bind, then try your test again and asterisk should see the SIP packets coming in.
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http://voisonics.com/
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admin at tootai.net
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| Posted: Wed Jan 22, 2014 8:41 am Post subject: [asterisk-users] Asterisk not receiving call from VPN addres |
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Le 20/01/2014 03:51, David Cunningham a écrit :
| Quote: | Hi,
We have a Kamailio and Asterisk cluster, both machines being on a real
103.x IP address and also on a 172.x OpenVPN address.
The problem is that when Kamailo receives a call from the VPN and
forwards it to the Asterisk server on it's 103.x address, Asterisk
never sees the call.
If Kamailio receives a call from the VPN and forwards the call to the
Asterisk server on it's 172.x address then it works. However, if the
call isn't from the VPN then forwarding it to the 172.x address
doesn't work. So basically the problem is going between the real
network and the VPN.
The question is, how can we make this work when calls are received on
either network on the Kamailio server and are forwarded to Asterisk?
Using ngrep on the Asterisk server we see that it does receive the
INVITE, but Asterisk's logging shows no sign it at all. We guess it's
a Linux networking issue rather than Asterisk's fault, but don't know
where to fix it. We do have net.ipv4.ip_forward = 1 on both the
Kamailio and Asterisk servers.
Thanks in advance for any help.
|
Hi David,
if you have a 11.7.0 version try to downgrade to 11.6.0.1 See my post
about similar issue with no registration of our intranet phones
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Daniel
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dcunningham at voisoni...
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| Posted: Wed Jan 22, 2014 5:39 pm Post subject: [asterisk-users] Asterisk not receiving call from VPN addres |
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Hello,
We tried Asterisk 1.8 and 1.6, but not yet Asterisk 11. We'll keep it in mind.
In the meantime we've decided to try a different network configuration instead, so the VPN network is separated from what Asterisk sees.
Thanks for all the advice given.
On 23 January 2014 00:42, Administrator TOOTAI <admin@tootai.net (admin@tootai.net)> wrote:
| Quote: | Le 20/01/2014 03:51, David Cunningham a écrit :
| Quote: | Hi,
We have a Kamailio and Asterisk cluster, both machines being on a real 103.x IP address and also on a 172.x OpenVPN address.
The problem is that when Kamailo receives a call from the VPN and forwards it to the Asterisk server on it's 103.x address, Asterisk never sees the call.
If Kamailio receives a call from the VPN and forwards the call to the Asterisk server on it's 172.x address then it works. However, if the call isn't from the VPN then forwarding it to the 172.x address doesn't work. So basically the problem is going between the real network and the VPN.
The question is, how can we make this work when calls are received on either network on the Kamailio server and are forwarded to Asterisk?
Using ngrep on the Asterisk server we see that it does receive the INVITE, but Asterisk's logging shows no sign it at all. We guess it's a Linux networking issue rather than Asterisk's fault, but don't know where to fix it. We do have net.ipv4.ip_forward = 1 on both the Kamailio and Asterisk servers.
Thanks in advance for any help.
|
Hi David,
if you have a 11.7.0 version try to downgrade to 11.6.0.1 See my post about similar issue with no registration of our intranet phones
--
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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David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 8063 9019 |
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