Hi, first of all, here's the architecture of what I'm trying to do : Public client --> Kamailio and there is a private freeswitch behind kamailio. I use rtpproxy to connect the client and the freeswitch that is used at media server. If a call is not answered from userA to userB and userB doesn't have a voicemail configured, I route the Invite to FreeSwitch and I change the destination user to 666666 that take care of playing a nice message that the user doesn't have a voicemail. My problem is that the SDP Connection doesn't get updated to the right IP. It get changed to the external IP instead of the internal. Here's the original INVITE Packet : 13:21:22.824290 IP 8.8.8.16.30893 > 8.8.8.18.5060: SIP: INVITE sip: 101(a)mydomain.com SIP/2.0 Eh....@.?...........x......mINVITE sip:101@mydomain.com SIP/2.0 Via: SIP/2.0/UDP 192.168.0.236:5060;branch=z9hG4bK-3e3b9e22 From: 102 <sip:102@mydomain.com>;tag=cacbdfac40ddd17ao0 To: <sip:101@mydomain.com> Remote-Party-ID: 102 <sip:102@mydomain.com>;screen=yes;party=calling Call-ID: 3ae5b914-591b8732(a)192.168.0.236 CSeq: 101 INVITE Max-Forwards: 70 Contact: 102 <sip:102@192.168.0.236:5060;ref=102> Expires: 240 User-Agent: Cisco/SPA122-1.3.5r(003) Content-Length: 263 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 16833207 16833207 IN IP4 192.168.0.236 s=- c=IN IP4 192.168.0.236 t=0 0 m=audio 16396 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv Note that 192.168.0.x in the internal network of the clients and 8.8.8.16 is the public IP of the client and 8.8.8.18 is the kamailio ip. Here's the INVITE packet sent from kamailio to Freeswitch 13:21:32.947196 IP 172.16.0.18.5060 > 172.16.0.19.5080: SIP: INVITE sip:666666666666666@172.16.0.19:5080 SIP/2.0 E... ...@. ...e...e.......P1INVITE sip:666666666666666@172.16.0. 19:5080 SIP/2.0 Record-Route: <sip:8.8.8.18;lr=on;ftag=cacbdfac40ddd17ao0;did=f3f.205> Test-Pascal: SDP avec freeswitch Via: SIP/2.0/UDP 172.16.0.18;branch=z9hG4bK31ea. e52b370f8248032680211fc7c253902a.1 Via: SIP/2.0/UDP 192.168.0.236:5060;rport=30893;received=8.8.8.16;branch=z9hG4bK-3e3b9e22 From: 102 <sip:102@mydomain.com>;tag=cacbdfac40ddd17ao0 To: <sip:101@mydomain.com> Remote-Party-ID: 102 <sip:102@mydomain.com>;screen=yes;party=calling Call-ID: 3ae5b914-591b8732(a)192.168.0.236 CSeq: 101 INVITE Max-Forwards: 69 Contact: 102 <sip:102@192.168.0.236:5060;ref=102;alias=8.8.8.16~30893~1> Expires: 240 User-Agent: Cisco/SPA122-1.3.5r(003) Content-Length: 279 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 16833207 16833207 IN IP4 8.8.8.18 s=- c=IN IP4 8.8.8.18 t=0 0 m=audio 46134 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=nortpproxy:yes Note that 172.16.0.18 is the internal IP of kamailio and 172.16.0.19 is the internal IP of FreeSwitch Here's the revealing par of kamailio configuration that us use when falling back to freeswitch : failure_route[TOVOICEMAIL] { $ru = "sip:666666666666666@" + $sel(cfg_get.freeswitch.bindip) + ":" + $sel(cfg_get.freeswitch.bindport); $du = "sip:666666666666666@"+ $sel(cfg_get.freeswitch.bindip) + ":" + $sel(cfg_get.freeswitch.bindport); force_send_socket(udp:172.16.0.18:5060); if (is_request()) { xlog("L_INFO","$ft -- NATMANAGE -- Request"); if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLT_NATS); } } } if(nat_uac_test("8")) { xlog("L_INFO","---------------------------------------------------"); xlog("L_INFO","---------------------------------------------------"); xlog("L_INFO","SDP Body avant : $fu --> $du | $ru"); xlog("L_INFO","---------------------------------------------------"); xlog("L_INFO","$ft -- RTPMANAGE(FreeSwitch) : NAT SDP"); rtpproxy_manage("1cowei",'192.168.101.18'); insert_hf("Test-Pascal: SDP avec freeswitch\r\n"); xlog("L_INFO","---------------------------------------------------"); } else { xlog("L_INFO","$ft -- RTPMANAGE(FreeSwitch) : PAS DE NAT SDP"); rtpproxy_manage("cowrei"); } if (is_reply()) { xlog("L_INFO","$ft -- NATMANAGE -- Reply"); if(isbflagset(FLT_NATS)) { if(is_first_hop()) set_contact_alias(); } if ( $si == $sel(cfg_get.freeswitch.bindip) && is_method("SUBSCRIBE") ) { remove_hf("Contact"); insert_hf("Contact: \$fU\ <sip:$fU@8.8.8.18:5060>\r\n","Expires"); } else { fix_nated_contact(); } } if ( is_method("INVITE") ) setflag(FLT_DIALOG); if (!t_relay()) { xlog("L_INFO","$ft -- ERROR Relay : $rc"); sl_reply_error(); } ------------------------------------------------ This configuration is working if the call is not going through failure_route I use Kamailio version 5.0.1 on Debian. Thank you Pascal _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users