<div dir="ltr"><div><div>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.<br>
<br>if(is_present_hf("P-Proxy-Route")){<br>            xlog("L_ERR","We have a Proxy Route request, performing loose routing to end point [$(hdr(Route))]");<br>            remove_hf("P-Proxy-Route");<br>
            remove_hf("Route");<br>            <br>            if(loose_route()){<br>                route(RELAY);<br>            }else{<br>                sl_send_reply("404","Unable to route request");<br>
            }<br>            <br>        }<br><br><br></div>Here is the inbound and outbound INVITE (I have removed hosts and IPs intentionally)<br><br></div>IN:<br><div><pre>INVITE <a href="http://sip:1177000012@dialer201.blah.com:5060">sip:1177000012@dialer201.blah.com:5060</a> SIP/2.0
Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK7495b309;rport
From: "+14109972688" <<a href="mailto:sip%3A%2B14109972688@blah.com">sip:+14109972688@blah.com</a>>;tag=as11488f8f
To: <<a href="http://sip:1177000012@dialer201.blah.com:5060">sip:1177000012@dialer201.blah.com:5060</a>>
Contact: <<a href="mailto:sip%3A%2B14109972688@0.0.0.0">sip:+14109972688@0.0.0.0</a>>
Call-ID: <a href="mailto:629cb67912fd16af758e6e7e67e89940@blah.com">629cb67912fd16af758e6e7e67e89940@blah.com</a>
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: <<a href="mailto:sip%3A1177000012@sipgateway.blah.com">sip:1177000012@sipgateway.blah.com</a>>
P-Proxy-Route: Yes
Content-Type: application/sdp
Content-Length: 242<br><br><br></pre><pre>OUT:<br>INVITE <a href="mailto:sip%3A1177000012@sipgateway.blah.com">sip:1177000012@sipgateway.blah.com</a> 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" <<a href="mailto:sip%3A%2B14109972688@blah.com">sip:+14109972688@blah.com</a>>;tag=as11488f8f
To: <<a href="http://sip:1177000012@dialer201.blah.com:5060">sip:1177000012@dialer201.blah.com:5060</a>>
Contact: <<a href="mailto:sip%3A%2B14109972688@0.0.0.0">sip:+14109972688@0.0.0.0</a>>
Call-ID: <a href="mailto:629cb67912fd16af758e6e7e67e89940@cf-dialer.com">629cb67912fd16af758e6e7e67e89940@cf-dialer.com</a>
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><br></pre><br></div></div>