[Asterisk-Users] Backend network, one-way audio...trunking

[Tim Connolly]

	I've got a cvs-head box running RHES4. I have it spread across two
vlans as I am using 802.1q trunking to my cisco switch. On the "front" side,
everything works great. If I move a phone from the front to the rear, change
its IP address and its config to reflect that move, and update the SIP
config, I end up with one-way audio. This server also runs vrrp, but I'm
using the real interface (vlan interface) as the destination of the sip.
I'll post a sip debug tomorrow if we can't figure this out...

Thanks
Tim