asterisk users - Aug 2004 - (no subject)

The wiki page on Asterisk + Nat
(http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions) lists the
possible types of server/client relationships with one most probably
interesting to us being #3.

<snip>
   3. Asterisk as a SIP server behind nat, clients on the outside
connecting to Asterisk
<snip>

Then it goes on to say:
*  #3 Works with port forwarding and some header mangling magic

Can somebody explain a little more about the "header mangling magic"
as
it is not discussed anywhere else in the document.

Currently I have my firewall port forwarding 5060 to my asterisk server
and the UDP port range forwarded as well. Registration works, but no
audio. Obviously the RTP stuff is not happy with the forwarding.

dbc.
--
David Cook

David Cook wrote:
> <snip>
>    3. Asterisk as a SIP server behind nat, clients on the outside
> connecting to Asterisk
> <snip>
> 
> Then it goes on to say:
> *  #3 Works with port forwarding and some header mangling magic
> 
> Can somebody explain a little more about the "header mangling
magic" as
> it is not discussed anywhere else in the document.
No magic required...
> Currently I have my firewall port forwarding 5060 to my asterisk server
> and the UDP port range forwarded as well. Registration works, but no
> audio. Obviously the RTP stuff is not happy with the forwarding.
In sip.conf ensure:

externip = <external ip>
localnet = 255.255.255.0
nat=yes

If it still doesn't work, enable "sip debug" at * CLI to
troubleshoot.
This will allow you to examine the sip headers to see why the phone & 
asterisk aren't talking.

Ryan