[Asterisk-Users] (no subject)

[Ryan Courtnage]

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