Okay so this is weird, I will explain the best I can.
I have everything setup and it was working great when I had it on an internal IP (192.168.1.*) locally. Then I moved the server outside my NAT and gave it a dedicated IP address and set it up all.
Now all my SIP phones register correctly, which is great, but there are a few problems:
extension to extension dialing just rings and rings then goes to voicemail and that person’s extension never actually rings.
If i call from outside into the system and dial an extension i hear it rining on my phone but the extension never rings and goes to voicemail.
Its as if it can’t route the calls, but I have no clue where to begin, any clue or any help would be appreciated?
Calls out work fine.
what you describe sounds like a typical NAT issue. read the sticky at the top of the forum for wiki pages that will help you.
can i ask why you wanted to move your Asterisk server outside the NAT ?
for several reasons:
we want to have external users connect
which is most important is we want it to be mroe of a central location as we have multiple offices.
What it looks like to me is that I may just have to add a box to each branch and then connect the boxes, which is not really what I wanted as I wanted it to be more of a ‘hosted’ situation.
you can probably fix this by moving your server back inside your firewall and setting up port forwarding for the ports that you need.
in rtp.conf, you can define the range of RTP ports to use…forward those along with 5060 (or whatever you define in sip.conf) and you should be good to go. if you’re natting on the far end as well, see if your endpoints support STUN.
i will say that having remote servers is good for a number of reasons, mainly because IAX is very easy to NAT with (one port does everything) and IAX trunking can save you quite a bit of bandwidth if you have multiple calls going down the same line…