Hi Tony.  I've got a similar problem.  4.0.2 to 4.0.4 works fine, 4.0.4
to 4.0.4 works fine.  DNS SRV records in place and fully resolvable.
Site-to-Site dial plan using sip domain names etc.  When I try 4.2 to
4.0.4 I experience the same problem as you. No NAT involved on a fully
routable LAN. I haven't dug into the detail as far as you, but if it is
any comfort, I think I can also reproduce this as well...

Dave B.
  

-----Original Message-----
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony
Graziano
Sent: May-19-10 9:20 AM
To: mel...@gmail.com
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Private dialing rules

I stopped at 4.2 dialing to 4.0.4. I've done 4.0.4 to 4.0.4 before
without
issue.

I plan to do some upgrades, one at a time, this stops be from doing some
work in production in 4.2 until the nat traversal is explained. Nat
traversal in 4.2 is being called only when I use the sipdomain as the
gateway. If I use hostname it works fine.
============================
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431

Email: tgrazi...@myitdepartment.net

LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427

Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/

----- Original Message -----
From: Melcon Moraes <mel...@gmail.com>
To: Tony Graziano <tgrazi...@myitdepartment.net>
Cc: Sipx-users list <sipx-users@list.sipfoundry.org>
Sent: Wed May 19 09:08:25 2010
Subject: Re: [sipx-users] Private dialing rules

How is the other way around behaving, as in a 4.0.4 box calling the
4.2.0
box?

-
MM

On Tue, May 18, 2010 at 14:39, Tony Graziano
<tgrazi...@myitdepartment.net>wrote:

> I am testing an ipsec connection and routing calls from a 4.0.4 system
and
> a 4.2 system via a site-to-site rule.
>
> Both my sipx systems can resolve each others DNS SRV records, ping and
> traceroute to each other via the ipsec connection.
>
> When initiating a call from 4.2 to the remote system, it fails with
"Too
> Many Hops". In doing a trace, I see the Record-Route: with the
internet ip
> address for my 4.2 system. When I created the unmanaged gateway, it
does
> not
> use the built-in sbc, so why is it getting involved with the call
routing?
>
> I have seen identical issues when using a sip trunk  or unmanaged
gateway.
> While I think this should be a simple configuration, I find that
> everything
> I do shows me that the built-in SBC is part of the record-route and
hence
> it
> tries to traverse nat.Why is that?
>
> Is what I am trying to do outside the design? It will work over the
native
> Internet traversing NAT in either direction, but it will bypass the
SBC
> for
> an ipsec connection. I don't understand.
>
>
> --
> ======================
> Tony Graziano, Manager
> Telephone: 434.984.8430
> sip: tgrazi...@voice.myitdepartment.net
>  Fax: 434.984.8431
>
> Email: tgrazi...@myitdepartment.net
>
> LAN/Telephony/Security and Control Systems Helpdesk:
> Telephone: 434.984.8426
> sip: helpd...@voice.myitdepartment.net
>  Fax: 434.984.8427
>
> Helpdesk Contract Customers:
> http://www.myitdepartment.net/gethelp/
>
> Why do mathematicians always confuse Halloween and Christmas?
> Because 31 Oct = 25 Dec.
>
>
> _______________________________________________
> sipx-users mailing list sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users
> Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
> sipXecs IP PBX -- http://www.sipfoundry.org/
>
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