Re: [Shorewall-users] SIP/RTP rules -- new to Shorewall

[Robert M. <rg...@ht...>]

Boy do I need to go out in the snow-covered streets for a walk to clear 
my head.

All the time my testing was dialing the WRONG number. Of course I was 
NOT seeing any traffic. Nothing in the Shorewall logs. Wireshark did not 
capture any packets from Broadvoice. Finally logged into my Broadvoice 
account, and it did not show any incoming calls. hmmm. Checked the logs 
in my cell phone (what I was using for the test), and my dyslexia bit me 
again.

So testing with the proper number and:

Simon Hobson wrote:
> Robert Moskowitz wrote:
>   
>> # SIP on UDP port 5060. Other SIP servers may need TCP port 5060 as well
>> ACCEPT  all-    all-    udp     5004:5082
>>     
>
> I would suggest limiting that to just dport 5060.
>
>   
>> ACCEPT  all-    all-    tcp     5060
>>     
>
> I've never heard of SIP using TCP
>
>   
>> # RTP - the media stream
>> ACCEPT  all-    all-    udp     10000:20000
>>     
>
> I would cut that down a bit - do you really need 10,000 simultaneous 
> call capacity ! Don't forget to alter /etc/asterisk/rtp.conf (IIRC) 
> to suit. Also, start at 10001 not 10000 as that is used for Webmin.
>   
call works just fine.

Now to add in the accounting.

But probably first to install the final production version of Trixbox 
2.4 that was released today!

Thanks for all of your help here.

I will probably be back in a few days!