[Asterisk-Users] Nat, SIP, Realtime problem

[Hall, Eric M.]

Just wanted to also say this does not happen to all users behind a NAT
box on RR or DSL line just a few.


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Tuesday, February 14, 2006 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Nat, SIP, Realtime problem

Hall, Eric M. wrote:

> Asterisk CVS-HEAD dated 2005-08-18
> WhitBox Linux respin 2
> mysql  Ver 11.18 Distrib 3.23.58
> Cisco 7960G
> 
> We are using the real-time drivers for sip and everything is working 
> great.
> They have a few employees that use the phones from home on a RR or DSL

> line.
> The problem is if they make a call everything works great they hang up

> and are able to get inbound calls. If they do not make a call for 5 or

> 10 mins they are unable to get inbound calls. If they dial out again 
> its all working for another 5 or 10 mins. This does not happen to all 
> remote people just a few.

Using Realtime SIP peers does not allow for "NAT Keepalive" packets to
be sent, so the firewall/NAT devices that those phones are connected to
are closing the SIP port hole after an expiration timeout.

To fix this, you'll need to upgrade to newer Asterisk (you really should
be running 1.2) and use 'realtime caching' for your SIP peers.
_______________________________________________
--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