[Asterisk-Users] SPA and NAT traversal

Jim Sturtevant jim.public at thesturtevants.com
Sat Apr 9 12:24:38 MST 2005


Thank you for your reply.  There is a wealth of information on the wiki,
etc.   I turned on RTP debug and the SPA is not sending it's public IP it is
sending it's NAT IP (192.168.1.100) so *'s RTP packets are going nowhere...


The SPA is behind a NAT and traversing the public IP network to get to the *
server.  It is successfully registering, thus I can ring a phone registered
locally to the * server.

I made sure localnet=192.168.2.9/255.255.255.0 (my local cfg for *)  and
externip=65.87.x.x (which is the public IP of my * server).  The * server is
behind a NAT as well with the 5060 and 16384-32767 UDP ports open.  

Based on RTP debug it appears the RTP packets are making it to the * server,
the problem is the return address is the internal NAT address of the SPA
192.168.1.100 and not it's public address.

Are you willing to share your Martha collection or are you going to keep it
to yourself? :-)


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric Wieling
Sent: Saturday, April 09, 2005 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SPA and NAT traversal

Jim Sturtevant wrote:

> I was hoping someone might help me diagnose a NAT issue with an SPA-2000
and
> my * server.  
> 
> My SPA is behind a NAT accessing a server which is also behind a NAT but
SIP
> and RTP ports are forwarded to it.
> 
> My SPA can successfully register.  It can call another extension which is
> inside the * local net and the inside phone can call the SPA.  But, no
> speech path either way.  I have NAT=YES and the two invite parameters are
> set to NO.

I'm desperately trying to get your sip.conf file telepathically but 
all I'm getting is images from your Martha Stewart porn collection. 
*shudder*

In addition to nat=yes you also need localnet= and externip= set, as 
shown in sip.conf.sample.


-- 
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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