I'm having a little problem and was wondering whether anyone would have any ideas or pointers for me.
I've been working on load-balancing asterisk and have had a pretty successful setup using LVS and IP tunneling (plus a bit of iptables nating).
I am only load balancing the SIP registration while the RTP between the SIP phone and the asterisk server and between the asterisk server and the CISCO AS5300 is being done directly with the real IP.
Now this setup worked wonderfully, and I had tested with SIP phones behind different routers to see if Natting wasn't causing a problem and everything worked fine.
But one of my locations recently changed routers (Linksys WRT54G)
and the SIP phone no longer registers with the asterisk servers.
After a bit of sniffing adn testing, here's what I came up with.
If the phone connects directly to the asterisk server without load-balancing, it works fine.
If the phone connects to the asterisk server through the load-balancing, the REGISTER packet comes into the asterisk server, but the reply instead of being sent-out from source-port 5060, it's sent out from source-port 1343 (or other lowest free port (1024,1026) and is blocked at the linksys gateway.
Any ideas why asterisk doesn't use the 5060 source port in the reply?
I'm unfortunately using version 0.9.0 of asterisk (my boss doesn't want to go with CVS).
P.S. The iptables part of the load-balancing NATs the source IP of the reply packets as being from the virtual IP because asterisk sets it as from the real IP. The rest is "normal" lvs
Thanks for any help Benjamin
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