On Wed, Mar 3, 2010 at 4:20 PM, Robert Joly <rj...@avaya.com> wrote: > > > > -----Original Message----- > > From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] > > Sent: Wednesday, March 03, 2010 4:16 PM > > To: Joly, Robert AVAYA (CAR:9D30) > > Cc: Sipx-dev list; Sipx-users list > > Subject: Re: [sipX-dev] sipxbridge and remote users without > > server behind nat > > > > > > > > On Wed, Mar 3, 2010 at 4:01 PM, Robert Joly <rj...@avaya.com> wrote: > > > > > > > Does sipxbridge have to be behind nat in order for remote > > > users or trunking to work? > > > > > > Yes as long as you uncheck the 'Server behind NAT' > > checkbox under > > System->Internet Calling->NAT Traversal. BTW, just to > > make sure, > > sipXbridge and remote NAT traversal are two different > > beasts - please > > see questions #1 and #2 of the FAQ section at the bottom of > > > > http://wiki.sipfoundry.org/display/xecsuserV4r0/Remote+User+NA > > T+Traversa > > l. > > > > 8< > > > > > > > > That was unchecked already. > > > > Remote User = Yes > > Server behind NAT = No > > > > All standard settings (static IP, not STUN), calls sent to port 5080. > > Are the phones also pointed at port 5080? 5080 is the port to be used > by ITSPs. All users, local or remote, must connect to port 5060. > > Phones and dns srv are for 5060 for sip. itsp sends calls to 5080.
I do this without much issue at all behind NAT. http://www.myitdepartment.net/support/sipx_bridge_pfsense_bandwidth-dot-com.pdf > > > > ITSP is from the template and setup just like my others with > > the same ITSP. > > > > User registers. Can call the AA,but not VM (no audio, the > > call connects). User can call out and be called. > > > > Has anyone ever done this before? > > I had a close variant of that, but not that setup exactly. > The thing that I find is that "without" the server behind NAT, it doesn't work like I would have thought, so I thought I'd ask if there was a limitation of some sort.
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