Interesting to hear that sipXecs works better for you when behind a NAT.
I would like to take a look at a network trace of your system to
understand why you are not getting the audio on VM calls. Can you run
'tcpdump -n -nn -s 0 -i any -w vm_no_audio.pcap' on your sipXecs,
reproduce the problem and send me the trace?
Thanks,
bob
________________________________
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Wednesday, March 03, 2010 4:25 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 4:20 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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