[Asterisk-Users] SIP behind NAT, workaround to make W Snel's very welcome fix work both for inside *and* outside clients

[Christopher Stephens]

Well, my hosts hack-on-hack didn't work...internal clients could register
with * using the hosts-hacked FQDN, and * could register with (for
example) FWD and iconnecthere, but on calls in either direction, I only
got a few seconds of audio, then silence (though debugging showed what
looked like a continued normal SIP/RTP conversation), then it threw a 484
Address Incomplete status and disconnected.

Would it be inappropriate to sponsor ($) a dev contest for the real
(universal/not hardcoded) 'hack'? :)


----- Original message -----
From: "Chris Albertson" <chrisalbertson90278 at yahoo.com>
To: asterisk-users at lists.digium.com
Date: Wed, 29 Oct 2003 09:13:31 -0800 (PST)
Subject: Re: [Asterisk-Users] SIP behind NAT, workaround to make W Snel's
very welcome fix work both for inside *and* outside clients


--- Peter Zeltins <peter at fintrading.com> wrote:
> > That's for pointing out Walter Snel "hack".
> > Adding his two additional features would not be
> > hard.
>
http://lists.digium.com/pipermail/asterisk-users/2003-October/024968.html
> 
> Any idea when these "hacks" will appear in CVS?

We should all hope "never".  That's why you call it a "hack"
because it works for only one very specific case and would break
SIP under Astrisk for most people.  It even breaks calls
between Asterisk and local SIP phones.

Now the trick is to write some code that desides if the trick is
to be used or not for each call by comparing the IP address of
Asterisk and the called SIP phone.

You migh want to experiment with it and report results.



=====
Chris Albertson
  Home:   310-376-1029  chrisalbertson90278 at yahoo.com
  Cell:   310-990-7550
  Office: 310-336-5189  Christopher.J.Albertson at aero.org
  KG6OMK

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