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
