Advanced routing question (SIP)

Hello, I can’t quite wrap my head around this one - So I am requesting input…here is my config:

-Asterisk PBX is on 192.168.1.x and is behing a nat’ed router (Obviously)
-I have a remote network of 10.10.40.x with a Grandstream GXT2000 also behind NAT
-The two net’s have a pin up VPN tunnel established (router to router)
-You can ping both ways between the nets to any device just fine

So my question is…do I need to mess with the NAT config on either side? I would think the tunnel would eliminate the need for any NAT transversal config, but I can’t seem to get the remote phone to register via sip?

-Could it be that asterisk won’t take registrations from the 10.10.40.x IP? The asterisk docs indicate that all private nets are considered local nets?

Anyway…any thoughts…anyone??


Do you have routes set in the Linux to say what the gateway is to the 10 network ? and have you the Localnets set correctly ?

you need in the server a route set to the 10 network and you also need the 10 network defined as a localnet


FYI (I am using *now)

In the GUI > SIP Config > Local Network Address is -

? How do I set two local nets in the box…separate with a comma?

I have set no static routes in Rpath…can you elaborate on this? I am familiar with windows routing and a noob to Linux…although I can SSH in and navigate Rpath, I have no idea the syntax to set a static route.

I assumed the router would be smart enough to handle it as the two nets can ping each other…the Localnet GW ( is the same physical device that is doing the tunnel…so technically they would have the same local gw (???). I would think that a packet hitting the router for the 10.10.40 net would get auto routed across the tunnel…but really I have no idea…that is the only way I can explain ping working from the 192 net to the 10 net.

FYI, the Gui help box says:

Local Network Address: ‘’ : All RFC 1918 addresses are local networks, ‘’ : Also RFC1918, ‘’ : Another RFC1918 with CIDR notation, ‘’ : Zero conf local network

so I assume that asterisk assumes the 10.10.40 net is local :wink:


They are Local networks as defined by RFCs not Localnet defined in asterisk.
to add another localnet you need to add a second line specifing it.

As to routes im not sure about rpath its rehat based so maybe similar. the route add command would I guess be the same as any other Linux.


Hi again, Here is the routeing table in rpath linux:

[root@PBX init.d]# route
Kernel IP routing table
Destination Gateway Genmask Flags Metric Ref Use Iface * U 0 0 0 eth0 UG 0 0 0 eth0 * U 0 0 0 eth0
default UG 0 0 0 eth0

and I added a second localnet line in sip.conf:


RIP V2 is enabled on both sides

Local router table:

Routing Table Entry List

Destination IP Address Subnet Mask Default Gateway Hop Count Interface * 40 ixp1 * 45 ipsec0 * 50 ixp0 10 ipsec0
default 40 ixp1

remote routing table:
Routing Table Entry List

Destination IP Address Subnet Mask Default Gateway Hop Count Interface 10 ipsec0 0 ixp0 * 50 ixp0 0 ixp1 * 40 ixp1 * 41 ipsec0
default 40 ixp1

frack…it all looks good to me…any other ideas?

Well, it looks like RIP was the trick…it just took a few minutes to register…but then i couldn’t make a call of any sorts…did a reboot of the phone, then it wouldn’t register?

Are there any NAT settings I need to worry about ya think?

tricky little tard…it will not beat me!

I thought this behavior was nothing to worry about, but now I think it may be a indication of something…

When I go to the remote IP phone (10.10.40.x) from the local network (192.168.1.x), the web interface starts to load, then hangs forever…like everything loads but the password box (the only input box) and the submit btn???

From the 10.10.40.x net, the web interface works as expected?

Now that is truly puzzling…

what is your phone using as it’s IP gateway?

The phone uses the same gw as the PC’s from DHCP…

Remote phone uses as the gw?

You have an idea g2110?

personally i would try to do some flood pings from both sides to determine if there is substantial packet loss going over the tunnel.

the fact that you got a partial load on the HTTP interface is indicative of some type of network/routing issue… at least IMO.

About 2500 ping’s 0% packet loss, between 40 and 50ms latency…that is fricking outstanding!

remote PC to local server and server to remote phone, both look the same?

I am no routing expert (maybe a routing pro), is it possible that there is a routing difference between ping and sip/UDP?

Things that make you go hmmmmmmmmm!

Does everyone agree that my config warrants NO NAT considerations? It seems to me that the tunnel would exempt me from needing nat translation as the two nets clearly know about each-other.

Another tidbit: Here is the tracert from local server to remote phone:


Tracing route to over a maximum of 30 hops

1 <1 ms <1 ms <1 ms
2 * * * Request timed out.
3 * * * Request timed out.
4 * * * Request timed out.
5 333 ms 442 ms 338 ms

Trace complete.

NOT what I expected…1 is the local router, 2,3,and 4?? One of those should be the remote router, then 5 is the remote phone. I wonder why they are not responding and why there are 3 in the middle (I would think there would be just one in the middle).

Ah ha! Tracepath (rpath traceroute) from PBX to phone fails at the local router ???

Here is the PBX routing table:

[root@PBX bin]# route
Kernel IP routing table
Destination Gateway Genmask Flags Metric Ref Use Iface * U 0 0 0 eth0 UG 0 0 0 eth0 * U 0 0 0 eth0
default UG 0 0 0 eth0
[root@PBX bin]#

The plot thickens…in fact, it turned black and is all gooey…little bit of smoke…