Can anyone tell me why Kamailio 4.0 is sending the INVITE out with the Route Header intact? The call is running through this block which I thought should remove it before sending it out. The far end is having a hard time dealing with the fact that the Route header is in there and we aren't looking for strict routing.
if(is_present_hf("P-Proxy-Route")){ xlog("L_ERR","We have a Proxy Route request, performing loose routing to end point [$(hdr(Route))]"); remove_hf("P-Proxy-Route"); remove_hf("Route");
if(loose_route()){ route(RELAY); }else{ sl_send_reply("404","Unable to route request"); }
}
Here is the inbound and outbound INVITE (I have removed hosts and IPs intentionally)
IN:
INVITE sip:1177000012@dialer201.blah.com:5060 SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK7495b309;rport From: "+14109972688" sip:+14109972688@blah.com;tag=as11488f8f To: sip:1177000012@dialer201.blah.com:5060 Contact: sip:+14109972688@0.0.0.0 Call-ID: 629cb67912fd16af758e6e7e67e89940@blah.com CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Aug 2013 23:27:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Route: sip:1177000012@sipgateway.blah.com P-Proxy-Route: Yes Content-Type: application/sdp Content-Length: 242
OUT: INVITE sip:1177000012@sipgateway.blah.com SIP/2.0 Record-Route: sip:0.0.0.0;lr;ftag=as11488f8f Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK2a77.07f36206.0 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK7495b309;rport=5060 From: "+14109972688" sip:+14109972688@blah.com;tag=as11488f8f To: sip:1177000012@dialer201.blah.com:5060 Contact: sip:+14109972688@0.0.0.0 Call-ID: 629cb67912fd16af758e6e7e67e89940@cf-dialer.com CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 16 Date: Wed, 14 Aug 2013 23:27:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 242 Route: sip:1177000012@dialer201.blah:5060