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:1177000...@dialer201.blah.com:5060 SIP/2.0
Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK7495b309;rport
From: "+14109972688" <sip:+14109972...@blah.com>;tag=as11488f8f
To: <sip:1177000...@dialer201.blah.com:5060>
Contact: <sip:+14109972688@0.0.0.0>
Call-ID: 629cb67912fd16af758e6e7e67e89...@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:1177000...@sipgateway.blah.com>
P-Proxy-Route: Yes
Content-Type: application/sdp
Content-Length: 242
OUT:
INVITE sip:1177000...@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:+14109972...@blah.com>;tag=as11488f8f
To: <sip:1177000...@dialer201.blah.com:5060>
Contact: <sip:+14109972688@0.0.0.0>
Call-ID: 629cb67912fd16af758e6e7e67e89...@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:1177000...@dialer201.blah:5060>
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