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 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


Thanks both…

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


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?


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 ???


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

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?

nat=always --> does exists always?? should be route?
callerid=device <6000>

[general] -->
Is this OK?

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

Thanks IANPLAIN![/b]