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 10.10.40.116 over a maximum of 30 hops
1 <1 ms <1 ms <1 ms 192.168.1.1
2 * * * Request timed out.
3 * * * Request timed out.
4 * * * Request timed out.
5 333 ms 442 ms 338 ms 10.10.40.116
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 192.168.1.1 ???
Here is the PBX routing table:
[root@PBX bin]# route
Kernel IP routing table
Destination Gateway Genmask Flags Metric Ref Use Iface
192.168.1.0 * 255.255.255.0 U 0 0 0 eth0
10.10.40.0 192.168.1.1 255.255.255.0 UG 0 0 0 eth0
169.254.0.0 * 255.255.0.0 U 0 0 0 eth0
default 192.168.1.1 0.0.0.0 UG 0 0 0 eth0
The plot thickens…in fact, it turned black and is all gooey…little bit of smoke…