Leif Madsen
2003-Sep-26 20:13 UTC
[Asterisk-Users] Creating a SIP gateway for use behind NAT
Hi all, Here is a graphical diagram of what I am trying to do: <SIP> <---> <GW/NAT/*> <--IAX--> <*> <--TDM400P--> <Analog Phone> So I have incoming SIP calls go to the * on the GW, which I then want to forward over IAX to the second * box behind the NAT GW. If I was to place a call on the second * box, it should then forward to the * on the NAT GW and place the call to the SIP destination. Thus, this should create a 2 way communication between the * box BEHIND the NAT GW, and end point. Now, the REASON I am doing this is because the TDM400P card I have is an older version, and has a lot of noise on the GW box, but is eliminated when using it in the box behind the GW. I realize I can just get this card replaced, but I am just using it on loan from the school, so until January comes around, I can't send it back, so this is my work around until then. I have built the IAX registrations between the boxes, so they register with each other. What I am trying to figure out is how to use my existing dial plan which worked on the gateway. This is going to work with FWD, so I should be able to receive calls on my 18924 number, and the 55555 welcome line as well. Couple of questions: 1) Should the GW or the second * box register with FWD? 2) Am I basically going to be using a blank configuration on the GW box, and having the second * box with all the fancy dial plan stuff, or the other way around? 3) Am I looking at using the switch => command? If so, an example would be fantastic, I've looked around on the digum list with google, but can't seem to find anything, perhaps I'm just using the wrong search words? Also, does my logic seem to make sense? Again, I would have just loved to put the TDM400P card in the GW machine, but unfortunately the dd line to make the CPU go to 100% just isn't very practical since I would like to use it for more than just *. If I can think if anything else, I will reply to this post. Thanks in advance for any help or direction, Leif Madsen.
Leif Madsen
2003-Sep-27 14:32 UTC
[Asterisk-Users] Creating a SIP gateway for use behind NAT
Leif Madsen wrote:> Hi all, > > Here is a graphical diagram of what I am trying to do: > > <SIP> <---> <GW/NAT/*> <--IAX--> <*> <--TDM400P--> <Analog Phone> > > So I have incoming SIP calls go to the * on the GW, which I then want to > forward over IAX to the second * box behind the NAT GW. If I was to > place a call on the second * box, it should then forward to the * on the > NAT GW and place the call to the SIP destination. Thus, this should > create a 2 way communication between the * box BEHIND the NAT GW, and > end point. > > Now, the REASON I am doing this is because the TDM400P card I have is an > older version, and has a lot of noise on the GW box, but is eliminated > when using it in the box behind the GW. I realize I can just get this > card replaced, but I am just using it on loan from the school, so until > January comes around, I can't send it back, so this is my work around > until then. > > I have built the IAX registrations between the boxes, so they register > with each other. What I am trying to figure out is how to use my > existing dial plan which worked on the gateway. This is going to work > with FWD, so I should be able to receive calls on my 18924 number, and > the 55555 welcome line as well. > > Couple of questions: > > 1) Should the GW or the second * box register with FWD? > 2) Am I basically going to be using a blank configuration on the GW box, > and having the second * box with all the fancy dial plan stuff, or the > other way around? > 3) Am I looking at using the switch => command? If so, an example would > be fantastic, I've looked around on the digum list with google, but > can't seem to find anything, perhaps I'm just using the wrong search words? > > Also, does my logic seem to make sense? Again, I would have just loved > to put the TDM400P card in the GW machine, but unfortunately the dd line > to make the CPU go to 100% just isn't very practical since I would like > to use it for more than just *. > > If I can think if anything else, I will reply to this post. >Hmmmmm, seems as if any sort of documentation or how to use switch is hard to find. If you know of a link, either on this list, or some external website, it would be very much appreciated. Thanks, Leif Madsen.
Leif Madsen
2003-Sep-27 15:09 UTC
[Asterisk-Users] Creating a SIP gateway for use behind NAT
OK.. lets just simplify this a bit. <remote> <--TDM400P--> <*> <--IAX--> <*> <--SIP--> <remote> I want to make a call from the very left asterisk box, through the right asterisk box to the remote end. So I have a phone plugged into the left * box, which is connected via an IAX connection to the right * box, which then places the call for me over SIP. I figure this should allow me to traverse NAT if the right * box is also on the GW machine (which I know works with SIP as it's just a firewall issue, and not NAT, which is simple). So, if someone can help explain how I can make a call from the left * box, through the right * box to the remote SIP connection, I'd be very grateful! I also promise to document anything and post it to the list for archival purposes, and on a website for an online reference. Thanks in advance, Leif Madsen.
Possibly Parallel Threads
- Forwarding SIP over IAX problem: No One Available
- Help with questions for initial Asterisk wizard (GUI)
- Does externalip= do anything to help with SIP behind a Linux based NAT router?
- Re: Asterisk behind NAT << How to do it. (Leif Madsen)
- Asterisk behind NAT << How to do it.