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


Michaƫl 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 <[EMAIL PROTECTED]> 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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