asterisk users - Jan 2006 - RE: Asterisk-Users Digest, Vol 18, Issue 131

Mark,

Thanks a lot for the feedback. It's reassuring to say the least

Mike



Message: 18
Date: Sat, 21 Jan 2006 15:36:18 -0500
From: Mark Phillips <g7ltt@g7ltt.com>
Subject: Re: [Asterisk-Users] SIP and NAT - best practices?
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users@lists.digium.com>
Message-ID: <43D29B42.3060705@g7ltt.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Most often the simple addition of nat=yes in the relevant sip.conf 
stanza is all that's required to make a remote SIP phone work from 
behind a firewall.

for example

[2201]
user=blah
secret=blah
auth=blah
allow=blah
host=dynamic
nat=yes

I've been running 4 remote SIP phones across the internet from my 
families houses all over the world in this manner. The only issues I get 
are those of bandwidth availability or rather occasional lack of it.

Hosted PBX's are no different. The hosting service should be providing a 
similar mechanism (although it might not be Asterisk based).

Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com


Michakl Gaudette wrote:
> Thanks Moises.  I was kind of hoping that, at least if I hosted my
Asterisk
> server somewhere where there was no NAT for the * box that the SIP phones
> wouldn't create any issues.  
> 
> How do you people with Hosted PBX handle the deployment of SIP phones
behind
> NAT firewalls? Is it just elbow grease and configuring every single phone
> for the customer, or is there a way?
> 
> Mike
> 
> 
> 
> you can redirect the ports of the router as well. Or you can configure
> your SIP phone to use a STUN server. Please read in voip-info.org
> about SIP NAT, there are good suggestions.
> 
> regards
> 
> On 1/20/06, Michakl Gaudette <michael.gaudette@virtutel.ca> wrote:
> 
>>Hello,
>>
>>I'm a bit new to SIP, and I've set up a SIP line with Asterisk
and my
>>wholesale provider.  That worked, fine.  I ahd to open up the ports on
my
>>router, forward them to the correct box, again fine.
>>
>>Now, if I get one of my customers to connect his SIP phone to my
Asterisk
>>box, and HE'S behind a NAT firewall, does he have to go through the
same
>>process, or is it just the Asterisk box that needs to translate the SIP
> 
> and
> 
>>RTP port?
>>
>>In other words: if my SIP phone is behind a Linksys router, do I need to
>>configure the Router for any reason?
>>
>>Mike
> 
> 
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