SIP Phone doesn't register trough ADSL

Hi all!

Just a simple question:

First, this is my scheme:

I have Asterisk running with an public IP. There is NO firewall.
I have an SIP Phone with a public IP (from the same pool that asterisk’s IP)
I have an SIP phone trough ADSL. This IP phone is behind a router, which have every ports opened.

Ok, this is what happens:

The SIP phone trough ADSL doesn’t register.
The SIP phone with a public IP do.

I can make calls from the SIP phone (ADSL) to SIP phone (public IP)
I can NOT make calls from the SIP phone (public IP) to SIP phone (ADSL)

The dialplan is OK, this is just a NAT problem?

sorry for my english,

Any ideas for this??
thanks in advance.

Maybe you will need to use a Stunt Server.
I think that there are some for free, but I not remember at the moment.
Just ask mr.google for Stunt protocol and Stunt Servers

:laughing: I think you mean Stun server. A Stunt server is a server that does the jobs a main server thinks are too dangerous. :wink:

But as to the question, if the server is on a real external address not in DMZ then it sounds like the localnet and externip arnt set correctly and make sure that the peer behind nat has nat=yes set

Ian

Thanks both…

[quote=“ianplain”]:make sure that the peer behind nat has nat=yes set

Ian[/quote]

I already made that, I test this thing with nat = yes and nat = no.

So, Stun Server you said?
can I install that in the same server that Asterisk?

Hi

Have you set the externip and localnet ?

try nat - always.

also do a sip debug while trying to register the set to see whats going on.

one question is how can the unregisterd phone make calls ???

Ian

[quote=“ianplain”]
one question is how can the unregisterd phone make calls ???

Ian[/quote]
In the phone setup, there’s an option for make calls without registration. So, asterisk accept this traffic (from this phone) and forward it.

The real question is: Why if I can do this calls, I can’t make it the otherwise.

Can I set externip on extensions definition?

Example:
[6000]
type=friend
secret=6000
qualify=yes
port=5060
nat=always --> does exists always?? should be route?
host=dynamic
dtmfmode=rfc2833
dial=SIP/6000
context=from-internal
canreinvite=yes
callerid=device <6000>

[general]
externip=extern-IP-phone.dyndns.org -->
Is this OK?

In a few minutes, I will post a SIP debug.

Thanks IANPLAIN![/b]