Thanks Klaus, the mhomed did the trick! I left the cfg file without listen parameter and works fine as well.
On Tue, Mar 23, 2010 at 5:42 AM, Klaus Darilion < klaus.mailingli...@pernau.at> wrote: > also set mhomed=1 > > http://sip-router.org/wiki/cookbooks/core-cookbook/devel#mhomed > > regards > kalus > > > Am 22.03.2010 21:00, schrieb Uriel Rozenbaum: > >> Hi gentlemen, >> >> >> I'm having a bit of a hard time developing a bridge for media sessions. >> >> I have a box with 2 interfaces; eth0 has the private IP p.r.i.v and eth1 >> has public IP p.u.b.l. >> >> When I receive a message in the private IP I try sending the call to a >> public proxy we have (p.u.b.2) using Carrierroute module but the source >> IP used is p.r.i.v. It is obvious that p.u.b.2 will not be able to route >> back messages. >> >> # >> U 192.168.200.11:5060 <http://192.168.200.11:5060> -> p.r.i.v:5060 >> >> INVITE sip:1568767...@p.r.i.v SIP/2.0. >> Via: SIP/2.0/UDP 192.168.200.11:5060;branch=z9hG4bK5d94bc09. >> From: "60911000" <sip:60911...@p.r.i.v>;tag=as6d477d46. >> To: <sip:1568767...@p.r.i.v>. >> Contact: <sip:60911...@192.168.200.11 <sip%3a60911...@192.168.200.11> >> <mailto:sip%3a60911...@192.168.200.11 <sip%253a60911...@192.168.200.11> >> >>. >> >> ... >> >> # >> U p.r.i.v:5060 -> p.u.b.2:5060 >> INVITE sip:1568767...@p.u.b.2 SIP/2.0. >> Record-Route: <sip:p.r.i.v;lr=on;ftag=as6d477d46>. >> Via: SIP/2.0/UDP p.r.i.v;branch=z9hG4bKf531.440788e1.0. >> Via: SIP/2.0/UDP >> 192.168.200.11:5060 >> ;rport=5060;received=192.168.200.11;branch=z9hG4bK5d94bc09. >> From: "60911000" <sip:60911...@p.r.i.v>;tag=as6d477d46. >> To: <sip:1568767...@p.r.i.v>. >> Contact: <sip:60911...@192.168.200.11 <sip%3a60911...@192.168.200.11> >> <mailto:sip%3a60911...@192.168.200.11 <sip%253a60911...@192.168.200.11> >> >>. >> >> ... >> >> I try then to set listen=0.0.0.0 so kamailio will be binded to all IPs >> but then the Route headers and Via use this 0.0.0.0 >> >> # >> U 192.168.200.11:5060 <http://192.168.200.11:5060> -> p.r.i.v:5060 >> >> INVITE sip:1568767...@p.r.i.v SIP/2.0. >> Via: SIP/2.0/UDP 192.168.200.11:5060;branch=z9hG4bK14a00df7. >> From: "60911000" <sip:60911...@p.r.i.v>;tag=as502405cf. >> To: <sip:1568767...@p.r.i.v>. >> Contact: <sip:60911...@192.168.200.11 <sip%3a60911...@192.168.200.11> >> <mailto:sip%3a60911...@192.168.200.11 <sip%253a60911...@192.168.200.11> >> >>. >> >> ... >> >> # >> U p.u.b.l:5060 -> p.u.b.2:5060 >> INVITE sip:1568767...@p.u.b.2 SIP/2.0. >> Record-Route: <sip:0.0.0.0;lr=on;ftag=as502405cf>. >> Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK711b.8d28549.0. >> Via: SIP/2.0/UDP >> 192.168.200.11:5060 >> ;rport=5060;received=192.168.200.11;branch=z9hG4bK14a00df7. >> From: "60911000" <sip:60911...@p.r.i.v>;tag=as502405cf. >> To: <sip:1568767...@p.r.i.v>. >> Contact: <sip:60911...@192.168.200.11 <sip%3a60911...@192.168.200.11> >> <mailto:sip%3a60911...@192.168.200.11 <sip%253a60911...@192.168.200.11> >> >>. >> >> ... >> >> In this case the destination UA cannot return the call because it does >> not have a valid IP to do so. >> >> Is there any way to make Kamailio act in a different way? Like letting >> me modify the IP it will use for Via and Record-Route headers. Be aware >> that I'm supposed to do some kind of mirror for calls coming from >> outside the network (origin is public and destination is private). >> >> I'm sure I'll be asking some more on this same thread as I test. >> >> Thanks in advance! >> Uriel >> >> >> >> _______________________________________________ >> Kamailio (OpenSER) - Users mailing list >> Users@lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> >
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