[SR-Users] Quick Question

[Klaus Darilion]

I also saw this recently in Kamailio and can not remember having this 
seen before.

In your case Kamailio is doing some loose-route/strict-route conversion 
which of course is buggy.

I think the problem is related that Kamailio does not correctly identify 
that the Route header addresses himself.

IIRC I solved it by changing the alias=.... section. I cant remember the 
details anymore, I played around with adding/removing the port from the 
alias command and suddenly it worked.

You see the "aliases" when you start Kamailio.

For debugging set debug=4 and inspect the log file for the INVITE. When 
entering loose_route() you see lots of "myself === ...." where Kamailio 
compares the Route header address with the local aliases (and listen 
sockets). I guess it matches (otherwise loose_route would fail). Then 
see if Kamailio maybe prints some log messages why it is doing 
loose/strict-routing conversion.

regards
Klaus

On 15.08.2013 01:31, Geoffrey Mina wrote:
> 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:
>
> INVITEsip:1177000012 at dialer201.blah.com:5060  <http://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 at blah.com  <mailto:sip%3A%2B14109972688 at blah.com>>;tag=as11488f8f
> To: <sip:1177000012 at dialer201.blah.com:5060  <http://sip:1177000012@dialer201.blah.com:5060>>
> Contact: <sip:+14109972688 at 0.0.0.0  <mailto:sip%3A%2B14109972688 at 0.0.0.0>>
> Call-ID:629cb67912fd16af758e6e7e67e89940 at blah.com  <mailto:629cb67912fd16af758e6e7e67e89940 at 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 at sipgateway.blah.com  <mailto:sip%3A1177000012 at sipgateway.blah.com>>
> P-Proxy-Route: Yes
> Content-Type: application/sdp
> Content-Length: 242
>
>
> OUT:
> INVITEsip:1177000012 at sipgateway.blah.com  <mailto:sip%3A1177000012 at 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 at blah.com  <mailto:sip%3A%2B14109972688 at blah.com>>;tag=as11488f8f
> To: <sip:1177000012 at dialer201.blah.com:5060  <http://sip:1177000012@dialer201.blah.com:5060>>
> Contact: <sip:+14109972688 at 0.0.0.0  <mailto:sip%3A%2B14109972688 at 0.0.0.0>>
> Call-ID:629cb67912fd16af758e6e7e67e89940 at cf-dialer.com  <mailto:629cb67912fd16af758e6e7e67e89940 at 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 at dialer201.blah:5060>
>
>
>
>
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