[SR-Users] Quick Question

[Carsten Bock]

Hi Geoffrey,

it looks like you add the Route header somewhere else, since the
Route-Header has a different position before and after Kamailio. Maybe
a lookup (with Path?) or similar? If you want to remove a header,
which you added previously in the routing logic, then you will have to
apply the changes before removing the header again
(msg_apply_changes())... better would be not adding the header at all.

Kind regards,
Carsten

2013/8/15 Geoffrey Mina <geoffreymina at gmail.com>:
> 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 at 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>;tag=as11488f8f
> To: <sip:1177000012 at dialer201.blah.com:5060>
> Contact: <sip:+14109972688 at 0.0.0.0>
> Call-ID: 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>
> P-Proxy-Route: Yes
> Content-Type: application/sdp
> Content-Length: 242
>
>
> OUT:
> INVITE sip:1177000012 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>;tag=as11488f8f
> To: <sip:1177000012 at dialer201.blah.com:5060>
> Contact: <sip:+14109972688 at 0.0.0.0>
> Call-ID: 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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>



-- 
Carsten Bock
CEO (Geschäftsführer)

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