[Asterisk-Users] Almost there--Remote connection

[Ferguson, Michael]

I made the firewall changes but still the same result.

On the GS100 phone, what us STUN server?
Why is it important?
If it say No in the config, I hear nothing.
If it says and has GS's STUN IP the connection is one way as noted
prior. Might this be the culprit?

Thanks... I am almost there....!! 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Deon
Rodden
Sent: Tuesday, October 19, 2004 11:35 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Almost there--Remote connection


My firewall script has something to the effect of:

# Allow Existing traffic through
-A INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT

# Incoming VOIP Ports
-A INPUT -m state --state NEW -m tcp -p tcp --dport 5036:5045 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 5036:5045 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 2727:2727 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 4569:4569 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 5060:5061 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 10000:20000 -j
ACCEPT

That's for IAX2 and SIP.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Karl Dyson
Sent: Tuesday, October 19, 2004 11:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Almost there--Remote connection

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> Ferguson, Michael
> Sent: 19 October 2004 16:18
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] Almost there--Remote connection
> 

[snip]

> 
> The * server is behind a firewall and I have opened ports 
> 10000-10100 5060
> 5004
> 4569
> 

IIRC, SIP uses 10000-20000 by default. Have you changed this to
10000-10100?

Cheers,

Karl

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