[Asterisk-Users] Why NAT problem

[Tom Rymes]

As a followup to my own post, AFAIK, my comments apply to SIP clients,
but you always have to forward the ports to the asterisk server...

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> Tom Rymes
> Sent: Saturday, August 13, 2005 10:22 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Why NAT problem
> 
> 
> This is not technically true. For instance, you can take a 
> Cisco 79X0 and put it behind NAT and it will work without 
> port forwarding. You do, however, have to program the phone 
> to enable the NAT features. (There are two, I can't remember 
> their names, though.) I have generally left the WAN IP 
> address blank, with no noticable ill effects, but that might 
> not be a good idea. 
> 
> Also, I believe that you can do this with multiple phones, so 
> long as you use different port numbers for each phone (5061, 
> 5062, etc)
> 
> Tom
> 
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> > Rudolf Ladyzhenskii
> > Sent: Saturday, August 13, 2005 9:53 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Why NAT problem
> > 
> > 
> > At firewall/NAT you have to do port forwarding.
> > 
> > If your phone is at port 5060, NAT device will receive a
> > connection and has 
> > to know that it is destined for your SIP phone. So, forward 
> > port 5060 to the 
> > phone.
> > 
> > Rudolf
> > 
> > 
> > ----- Original Message -----
> > From: "Kamran Ahmad" <p_kami at yahoo.com>
> > To: <asterisk-users at lists.digium.com>
> > Sent: Sunday, August 14, 2005 6:52 AM
> > Subject: [Asterisk-Users] Why NAT problem
> > 
> > 
> > > hello
> > >
> > > i am using asterisk-1.0.9. i have a NAT problem.
> > > without NAT registration is ok. and if user is bhind
> > > NAT it is registring on asterisk. but SJPhone is
> > > showing "not registered". i think asterisk is properly
> > sending request
> > > to UA. any comments............this sip.conf setting was working
> > > previously
> > >
> > >   -- Registered SIP '5000' at 0.0.0.0 port 5060
> > > expires 120
> > >    -- Saved useragent "SJLabs-SJphone/1.40.258" for
> > > peer 5000
> > >
> > > [general]
> > > context=default
> > > port=5060
> > > bindaddr=0.0.0.0
> > > srvlookup=yes
> > > nat=yes
> > > canreinvite=no
> > >
> > > [5000]
> > > type=friend
> > > port=5060
> > > canreinvite=no
> > > host=dynamic
> > > nat=yes
> > > insecure=yes
> > > auth=plaintext
> > >
> > >
> > >
> > >
> > >
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> 
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