I think my problem is numbrer 3 cause basicly my friend who is not on my
router is trying to get connected to me but can't and I am the 1 that is
behind a nat.
thanks
hank
----- Original Message -----
From: "Sebastian Silva" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[email protected]>
Sent: Tuesday, June 28, 2005 12:45 PM
Subject: Re: [Asterisk-Users] How do you handle NAT?
Hi everyone.
1. Asterisk as a SIP client behind nat, connecting to outside SIP
Proxies:
#1 works with a NAT-supporting proxy as SIP Express router as the outside
proxy. (Get an account at IPtel.org and try!). Fails with Free World
Dialup.
2. Asterisk as a SIP client behind nat, connecting to inside SIP proxies:
#2 Works- no NAT in between
3. Asterisk as a SIP server behind nat, clients on the outside connecting
to Asterisk:
#3 Works with port forwarding and some header mangling magic
4. Asterisk as a SIP server behind nat, clients on the inside connecting
to Asterisk:
#4 Works - no NAT in between
5. Asterisk as a SIP client outside nat, connecting to outside SIP
proxies:
#5 is no problem. No NAT in the middle
6. Asterisk as a SIP client outside nat, connecting to inside SIP proxies:
#6 is a problem if no port forwarding is done, similar to 3 above.
7. Asterisk as a SIP server outside nat, clients on the outside connecting
to Asterisk:
#7 is no problem. No NAT in the middle
8. Asterisk as a SIP server outside nat, clients on the inside connecting
to Asterisk:
#8 is solved with nat=yes and qualify=xxx in sip.conf for the client in
most cases. Some clients (X-lite) assist themselves by using STUN and
sending UDP keep-alive packets. Qualify sends keep-alive packets from
Asterisk to the client on the inside.
from wiki
Now, if you net to define a NAT, you have to set asterisk to
"canreinvite=no", "qualify=yes" and "nat=1".
Also, INSTEAD of NAT, you can use a STUN server. To use a STUN server you
should set asterisk to "canreinvite=no", "qualify=no" and "nat=0" (the
STUN configuration is in your agents).
Sebas
hank wrote:
how easy is it to set up a stun server? with asterisk amd will this fix
part of the nat problem?
----- Original Message ----- From: "Ray Van Dolson"
<[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[email protected]>
Sent: Tuesday, June 28, 2005 8:14 AM
Subject: Re: [Asterisk-Users] How do you handle NAT?
We've been feeling our way along with the NAT stuff (using SIP) as well.
At this point we are fairly small, so the keep-alive packets are not too
bad.
What type of user load are you at and what are the specs on your
Asterisk box?
I'm concerned we may run into this as well.
We do have the luxury that each Sipura device we use is sitting behind
its own
NAT (a customer CPE). So we can do port-forwarding and in combination
with a
STUN server (MyStun), things work quite well. The only issues left to
deal
with are a lingering problem with ip_conntrack entries staying cached
because
of the "keep alive" packets due to qualify=yes after the CPE's IP
address
changes.
Curious to hear other's setups as well. I would *love* to start using
the
IAXy instead, but it has a couple shortcomings over the Sipura 2002's
we're
using now:
- About $10/more
- Only has one line (apparently two lines is a bit more of a selling
point).
Still trying to figure out a good way to make a case for the IAXy
though.
Ray
On Tue, Jun 28, 2005 at 09:59:49AM -0500, Matthew Boehm wrote:
We are interested in how other people are handling NAT problems. We
have
several customers all of which have some sort of firewall/NAT device at
their location. For simplicity sake, all customers' internal networks
are 192.168.*.*.
Our asterisk box is on public IP not blocked by any FW/NAT.
I use QUALIFY=yes on all our customers' phones and I feel that sending
out 80-something keep-alive packets is causing our box to crawl and
cause bad calls.
Would SER be better in this case? Should I have phones register with
SER
instead of with Asterisk?
Thanks,
Matthew
P.S. Yes, I have read stuff on NAT on the wiki. I'm more interested in
other real world, working, solutions.
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--
Sebastian Silva
G R U P O G A U S S
Depto. Sistemas
Av. Libertador 6250 4 piso
Tl.: 4 706-2222 (int. 121)
[EMAIL PROTECTED]
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