Apart of what everyone writes with the NAT=YES I would suggest using
canreinvite=no as well as normally asterisk cans the reinvite and this
might cause the audio not to get through the NAT and cause dead air for
the users specially if the users are behind 2 seperate NAT servers eg.
different private networks.
By using canreinvite=no and nat=yes most of the NAT problems go away.
In this scenario the example would look like this:
[2201]
user=blah
secret=blah
auth=blah
allow=blah
host=dynamic
*nat=yes
canreinvite=no*
Mark Phillips wrote:
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
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users