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<DIV><FONT face=Arial size=2>Hello, NAT/Firewall is truely a problem in the
ITSP arena. There is one solution I know of that works
well as an integrated DHCP/NAT/Firewall into a SIP aware
firewall. Check out </FONT><A href="http://www.intertex.se"><FONT
face=Arial size=2>www.intertex.se</FONT></A><FONT face=Arial
size=2> </FONT> and look at the IXX66 products. They even have a
device that integrates DSL/NAT/Firewall. Or, one can purchase a SIP device
that supports STUN(Grandstream and SNOM are the only vendors I know of that do)
and install a STUN server. If anyone is interested I have a STUN server
running to test with. Hope this helped....</DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>Mike</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Michael Kane<BR>To-Talk Communications LLC.<BR>37
Sandusky Dr.<BR>Wareham, Ma. 02571<BR>508-295-2826</FONT></DIV>
<DIV><FONT face=Arial size=2>----- Original Message ----- </FONT>
<DIV><FONT face=Arial size=2>From: "John Todd" <</FONT><A
href="mailto:jtodd@loligo.com"><FONT face=Arial
size=2>jtodd@loligo.com</FONT></A><FONT face=Arial size=2>></FONT></DIV>
<DIV><FONT face=Arial size=2>To: <</FONT><A
href="mailto:asterisk-users@lists.digium.com"><FONT face=Arial
size=2>asterisk-users@lists.digium.com</FONT></A><FONT face=Arial
size=2>></FONT></DIV>
<DIV><FONT face=Arial size=2>Sent: Tuesday, July 01, 2003 3:47 PM</FONT></DIV>
<DIV><FONT face=Arial size=2>Subject: Re: [Asterisk-Users] A solution for SIP
and NAT</FONT></DIV></DIV>
<DIV><FONT face=Arial><BR><FONT size=2></FONT></FONT></DIV><FONT face=Arial
size=2>> I'm uncertain why you're not able to get SIP working for your user
<BR>> agents (SIP clients.) With Cisco equipment, as an example, it
works <BR>> quite well and almost every 79xx or ATA-186 I have is behind a
NAT, <BR>> and this configuration is duplicated across a dozen or more
systems <BR>> now running behind almost every conceivable NAT/PAT
situation*<BR>> <BR>> Known working config:<BR>> <BR>> UA ->
(NAT) -> Internet -> Asterisk<BR>> <BR>> Can you be more specific
about your problems with SIP? Perhaps you <BR>> have done so in the
past, but re-state and maybe someone can see what <BR>> the problem
is.<BR>> <BR>> JT<BR>> <BR>> <BR>> *Note: the Cisco PIX, while
supposedly SIP-friendly, has been the one <BR>> box that has not worked with
NAT/PAT SIP sessions. I have not been <BR>> the admin on that system,
but a fairly clueful Cisco wrangler has <BR>> been unable to make it work for
originating calls in both directions <BR>> - only one-way origination
works.)<BR>> <BR>> <BR>> >Hi all.<BR>> ><BR>> >I have
come to the conclusion that there just isn't anything out there<BR>> >for
allowing SIP and NAT to work together nicely. This is rather amazing<BR>>
>considering that as far back as March 2000 there are documents<BR>>
>describing how to do it.<BR>> ><BR>> >So I've started a really
simple SIP and RTP proxy project, SaRP, on<BR>> >sourceforge.net.
Yesterday we uploaded 0.2 of the perl based release.<BR>> >This is the
first general release and should work for most people. We<BR>> >are using
it quite successfully for standard calls between all sorts of<BR>> >NATed
clients. All you need to do is forward UDP/5060 from your<BR>>
>firewall/router to the box running SaRP if you want incoming calls
to<BR>> >work and also allow UDP traffic from the ports listed in the
config file<BR>> >out.<BR>> ><BR>> >The project can be found
at </FONT><A href="http://sarp.sourceforge.net/"><FONT face=Arial
size=2>http://sarp.sourceforge.net/</FONT></A><BR><FONT face=Arial size=2>>
><BR>> >I would be very interested in any feedback you may
have.<BR>> ><BR>> >Regards<BR>> ><BR>> >Andrew
Radke.<BR>> >_______________________________________________<BR>>
>Asterisk-Users mailing list<BR>>
>Asterisk-Users@lists.digium.com<BR>>
>http://lists.digium.com/mailman/listinfo/asterisk-users<BR>> <BR>>
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