Re: [asterisk-users] Help with IP Routing
Assume previous IP is LAN. Forwarding public IP ports to LAN is straighforward. However, with SIP headers you will (don't know H323) have to modify outgoing SIP headers: replace LAN ip with WAN ip. For callers you have to substitute RTP destination IP For callees you have to substitute RTP source IP. I`m afraid you will have to check more details here: http://www.ietf.org/rfc/rfc3261.txt Maybe client sends server it's own IP address? However, dumb header substitution + port range forwarding should work in all cases for SIP. On Wed, May 26, 2010 at 3:55 PM, Nivin Kumar nivinkuma...@yahoo.in wrote: Hello, I'm in a bit of a fix. We have a particular Windows based softswitch which is has its SIP and H323 ports hardcoded to listen on a particular IP address. The problem is that the ISP is having major issues and we can no longer depend on them for service. The softswitch will not listen on any other IP address and this can not be fixed. I was thinking of creating a NAT network wherein we will forward all traffic from another public ip address to this server, however I'm not sure how this will work. Do I need to modify the sip headers? Any thoughts or suggestions? Thanks, Nivin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with IP Routing
Try a Cisco ASA. It will rewrite the headers if configured properly. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Motiejus Jakštys Sent: 26 May 2010 14:17 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with IP Routing Assume previous IP is LAN. Forwarding public IP ports to LAN is straighforward. However, with SIP headers you will (don't know H323) have to modify outgoing SIP headers: replace LAN ip with WAN ip. For callers you have to substitute RTP destination IP For callees you have to substitute RTP source IP. I`m afraid you will have to check more details here: http://www.ietf.org/rfc/rfc3261.txt Maybe client sends server it's own IP address? However, dumb header substitution + port range forwarding should work in all cases for SIP. On Wed, May 26, 2010 at 3:55 PM, Nivin Kumar nivinkuma...@yahoo.in wrote: Hello, I'm in a bit of a fix. We have a particular Windows based softswitch which is has its SIP and H323 ports hardcoded to listen on a particular IP address. The problem is that the ISP is having major issues and we can no longer depend on them for service. The softswitch will not listen on any other IP address and this can not be fixed. I was thinking of creating a NAT network wherein we will forward all traffic from another public ip address to this server, however I'm not sure how this will work. Do I need to modify the sip headers? Any thoughts or suggestions? Thanks, Nivin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with IP Routing
Is there a tool that will allow me to automatically change sip headers in realtime? --- On Wed, 26/5/10, Motiejus Jakštys desired@gmail.com wrote: From: Motiejus Jakštys desired@gmail.com Subject: Re: [asterisk-users] Help with IP Routing To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, 26 May, 2010, 1:17 PM Assume previous IP is LAN. Forwarding public IP ports to LAN is straighforward. However, with SIP headers you will (don't know H323) have to modify outgoing SIP headers: replace LAN ip with WAN ip. For callers you have to substitute RTP destination IP For callees you have to substitute RTP source IP. I`m afraid you will have to check more details here: http://www.ietf.org/rfc/rfc3261.txt Maybe client sends server it's own IP address? However, dumb header substitution + port range forwarding should work in all cases for SIP. On Wed, May 26, 2010 at 3:55 PM, Nivin Kumar nivinkuma...@yahoo.in wrote: Hello, I'm in a bit of a fix. We have a particular Windows based softswitch which is has its SIP and H323 ports hardcoded to listen on a particular IP address. The problem is that the ISP is having major issues and we can no longer depend on them for service. The softswitch will not listen on any other IP address and this can not be fixed. I was thinking of creating a NAT network wherein we will forward all traffic from another public ip address to this server, however I'm not sure how this will work. Do I need to modify the sip headers? Any thoughts or suggestions? Thanks, Nivin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with IP Routing
Hello, if the remote side (the public IP side) is capable to do something like asterisk's nat=yes (in sip.conf), than a mascerading router (like every cheap DSL router) would do enough NAT do let SIP work. If the remote side does not support that nat-hack (which is not SIP standard), than you will need a NATing router also doing a lot of SIP header rewriting. Maybe the most easy thing will be to install asterisk on the NATing machine and operating regular SIP links on both sides. Roger. Nivin Kumar schrieb: Hello, I'm in a bit of a fix. We have a particular Windows based softswitch which is has its SIP and H323 ports hardcoded to listen on a particular IP address. The problem is that the ISP is having major issues and we can no longer depend on them for service. The softswitch will not listen on any other IP address and this can not be fixed. I was thinking of creating a NAT network wherein we will forward all traffic from another public ip address to this server, however I'm not sure how this will work. Do I need to modify the sip headers? Any thoughts or suggestions? Thanks, Nivin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with IP Routing
- Nivin Kumar nivinkuma...@yahoo.in wrote: Hello, I'm in a bit of a fix. We have a particular Windows based softswitch which is has its SIP and H323 ports hardcoded to listen on a particular IP address. The problem is that the ISP is having major issues and we can no longer depend on them for service. The softswitch will not listen on any other IP address and this can not be fixed. I was thinking of creating a NAT network wherein we will forward all traffic from another public ip address to this server, however I'm not sure how this will work. Do I need to modify the sip headers? Any thoughts or suggestions? --- And this is related to Asterisk.. how? If your 'particular Windows based softswitch' doesn't in fact allow you to change the listening interfaces then it sounds like one great piece of software. If you're going to post something completely OT to the list, at least have the courtesy of telling us what softswitch you're talking about? --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with IP Routing
Nivin Kumar schrieb: Is there a tool that will allow me to automatically change sip headers in realtime? Hi, imho changing the SIP headers will not be sufficient, since the old IP addresses are now private IP addresses (only in your network, outside, there are still public, but pointing not to your equipment). You will need a gateway, which does both: NAT 1:1, old IP addresses - new IP addresses and rewriting or all SIP headers, including those headers concerning the RTP endpoints. Maybe, you can do this with OpenSIPS. But I'm not sure about the SIP-headers for RTP. For H.323, it is imho less complicate, since it is robust for NAT and has no headers including IP addresses. Regards, Roger. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with IP Routing
Skip the whole NAT scenario. Put up an asterisk box with two network interfaces. One interface connects to the real world on your new IP address from your new ISP. The other interface can be on the same subnet as the windows box that you can't change. Set up a SIP trunk to your Windows box. Use packet 2 packet bridging in asterisk. Now that the emergency is over you can migrate off of your Windows thing at a more comfortable pace. You will be using someone else's public IP privately for awhile, but the main thing affected by that is your asterisk box won't be able to talk to anybody in that subnet in the outside world. You'll have to determine how bad of a thing that would be. BTW: What the heck is this software? Sounds like whoever wrote that wasn't thinking ahead. Hello, I'm in a bit of a fix. We have a particular Windows based softswitch which is has its SIP and H323 ports hardcoded to listen on a particular IP address. The problem is that the ISP is having major issues and we can no longer depend on them for service. The softswitch will not listen on any other IP address and this can not be fixed. I was thinking of creating a NAT network wherein we will forward all traffic from another public ip address to this server, however I'm not sure how this will work. Do I need to modify the sip headers? Any thoughts or suggestions? Thanks, Nivin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
