[Gllug] VoIP

Jack Richards jack at gnu-solutions.com
Sun Jan 9 11:10:26 UTC 2005


We build asterisk boxes quite a lot and though firewalls can be tricky, they 
are not as big a problems as most make out.

With SIP, a register command to the VOIP provider usually gives the IP address 
of the system that Asterisk is running on. Obviously, if you are running this 
on a NATed machine, this can cause problems. 

If using NAT, the best thing to do on the firewall is port forward ports 5060 
and 5080 and a a small range from 10,000 upwards ( user definable ) to the box 
behind the firewall runing NAT and specify in the Aserisk sip.conf the IP 
address of the firewall as the IP address to register with VOIP provider.

Asterisk works with a single IP address, the user handsets, can either use new 
IP addresses ( routable to the Asterisk box ) or if you are using PC phones, 
will co-exist with the PC they are attatched to.

Stear clear of Grandstream phones though.  The firmware is buggy as hell.  My 
colleagues use Sipura kit in most implementations where a standalone device is 
needed and the SIP100 and 3000 are very good devices. 

Also be carefull about non Digium FX0 cards as they do not match the UK line 
impedance and give all manner of echo, level and quality issues - found this 
out the hard way !

Jack


> > 
>    It may do the job, and well. It would mean that the VoIP and ethernet are
> entirely separate, the kind of approach taken by Solwise. I have seen a
> cautionary note about firewalls and NAT. This could be a big problem, or is
> it one that is easily circumvented? Does it require an IP block, or can it
> be made to work with a single IP address? The people who want the system are
> making large numbers of expensive international calls.
> 
> 
> -- 
> Chris Bell
> 
> -- 
> Gllug mailing list  -  Gllug at gllug.org.uk
> http://lists.gllug.org.uk/mailman/listinfo/gllug
> 


Jack Richards

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