I did a bit of digging and found the transformation mask thread and looked a bit more in the manual. and put:
00: 0:44 8:4416208
in my mask. This works but:
00: 0:44 8:016208
doesn't.
Probably because your carrier is looking for E164 numbering - hence the 44 requirement.
So do I really need to uninstall and reinstall all the rest of SAIL?
As long as you can modprobe zaptel and ztdummy without error, then no you don't need to re-install.
Wow, complicated little network.....
When a SAIL box is in server-only mode is it then unable to have remote phones register with it?
Not quite but it makes life harder. As a general rule, you can cross one firewall with SIP if you use symmetrical RTP. You can't cross two without using a session border controller or media proxy (which both parties can "see").
It might help put this into perspective for you if you read this....
http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutionsIt isn't entirely accurate but it gives a good general overview of what is and isn't possible.
The big "trick" for SIP where natted firewalls are concerned, is the use of Symmetrical RTP. SAIL uses the inbuilt asterisk RTP symmetry whenever you specify a phone as "remote" when you define it to SAIL. For all practical purposes, "remote" means "not on this subnet".
So, are you running your phones as "remotes" or "locals"?
Next point, we often run the SARK/SAIL box behind a Natted firewall but we still run it in server-gateway mode. This allows us to run all of the phones on their own sub-net (below the SARK box) and it works pretty well, except for remote phones, unless SARK is in the DMZ and receiving all SIP & RTP packets from the router/firewall, in which case remote phones work fine.
You will find in these circumstances that Wireshark (formerly ethereal) becomes your best pal. In many cases it is the ONLY way to see what is actually going on in the network and where the packets are flowing, and more importantly, not flowing.
Phones in DMZ initially register but are then "unreachable" and can call phones in private LAN but there is no sound.
This is usually because you have the
external ip address in Globals either not set or incorrectly set. It should reflect the IP address at your network border, i.e. the router/gateway. It is only relevant in server-only mode and asterisk uses it as the return address and puts it into all of the outbound packets. If it is set to an incorrect value (for example 192.168.1.100 - which is meaningless outside of the local subnet) then you will effectively "spoof" the target (phone) with the wrong return address.
Read the stuff on SIP NAT first so you have a feel for the art of the possible and then experiment by running tethereal at your SAIL box to see what packets are arriving and departing. Usually this will be enough to pinpoint where the failures are.
Kind Regards
Selintra
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