[OpenSIPS-Users] PRACK, 404 not here

[Răzvan Crainea]

Hi, Miha!

It looks like loose_route() fails - did you try to look into the logs 
and see if it indicates something? Is the SBC_1 IP advertised in the 
Route header a listener of OpenSIPS?

Also, if loose_route() fails, you should still try to match the PRACK 
against the dialog. So your scripting logic should look like this:

if (has_totag()) {
     if (loose_route() || match_dialog()) {
         ...
     }
}

Let us know how that goes.

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 09/07/2016 10:38 AM, Miha wrote:
> Hi
>
> i have one issue and do not know how to solve it...
>
> Initial invite:
>
> U SBC_2:5060 -> SBC_1:5060 INVITE 
> sip:777774220000 at SBC_1:5060;user=phone SIP/2.0. Via: SIP/2.0/UDP 
> SBC_2:5060;branch=z9hG4bK57fa.67ccbb16.0. From: 
> <sip:8888818100100 at PBX;user=phone>;tag=*1875283502*. To: 
> <sip:777774220000 at SBC_2;user=phone>. Call-ID: 
> *fb9e258ae909d311a85a0090332e03ed at PBX.* CSeq: 1193456 INVITE. Contact: 
> <sip:SBC_2;did=8d9.43418513>. Alert-Info: <urn:alert:source:internal>. 
> Allow: REGISTER,SUBSCRIBE,NOTIFY,INVITE,ACK,P
> Seq....U PBX:5060 -> SBC_2:5060 PRACK sip:SBC_2;did=8d9.43418513 SIP/2.0. 
> Via: SIP/2.0/UDP PBX:5060;branch=z9hG4bK-002AF6E3;rport. From: 
> <sip:8888818100100 at PBX;user=phone>;tag=*1875283502*. To: 
> <sip:777774220000 at SBC_2;user=phone>;tag=*FamBBcayZeKgF*. 
> Call-ID:*fb9e258ae909d311a85a0090332e03ed at PBX.* CSeq: 1193457 PRACK. 
> Content-Length: 0. Max-Forwards: 70. RAck: 1601153264 1193456 INVITE. 
> . Seq.... U SBC_2:5060 -> SBC_1:5060 PRACK 
> sip:777774220000 at SBC_1:5060;transport=udp SIP/2.0. Route: 
> <sip:SBC_1;lr;ftag=1875283502;did=8d9.e2509d35>. Via: SIP/2.0/UDP 
> SBC_2:5060;branch=z9hG4bK67fa.e6a41de2.0. From: 
> <sip:8888818100100 at PBX;user=phone>;tag=*1875283502*. To: 
> <sip:777774220000 at SBC_2;user=phone>;tag=*FamBBcayZeKgF*. Call-ID: 
> *fb9e258ae909d311a85a0090332e03ed at PBX.* CSeq: 1193457 PRACK. 
> Content-Length: 0. Max-Forwards: 69. RAck: 1601153264 1193456 INVITE.
>
> Seq....
> U SBC_1:5060 -> SBC_2:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 
> SBC_2:5060;branch=z9hG4bK67fa.e6a41de2.0. From: 
> <sip:8888818100100 at PBX;user=phone>;tag=1875283502. To: 
> <sip:777774220000 at SBC_2;user=phone>;tag=FamBBcayZeKgF. Call-ID: 
> fb9e258ae909d311a85a0090332e03ed at PBX. CSeq: 1193457 PRACK. Server: 
> OpenSIPS (2.1.1 (x86_64/linux)). Content-Length: 0.
>
> Why I am getting 404 from Opensips. is should be routed like seq request, right?
>
>
>
>
>
> if (has_totag()) { # sequential requests within a dialog should # take 
> the path determined by record-routing if (loose_route()) { 
> xlog("loose_route"); #if ($DLG_status!=NULL) xlog("dlg_status"); if 
> (!validate_dialog()){ fix_route_dialog(); xlog("fix_route_dialog"); } 
> if (is_method("BYE")) { setflag(1); # do accounting ... 
> #setflag(ACC_FAILED); # ... even if the transaction fails } else if 
> (is_method("INVITE")) { # even if in most of the cases is useless, do 
> RR for # re-INVITEs alos, as some buggy clients do change route set # 
> during the dialog. record_route(); xlog("check_fraud"); } # route it 
> out to whatever destination was set by loose_route() # in $du 
> (destination URI). route(relay); } else { if ( is_method("ACK") ) { if 
> ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an 
> ACK after # a 487 or e.g. 404 from upstream server t_relay(); exit; } 
> else { # ACK without matching transaction -> # ignore and discard 
> exit; } } sl_send_reply("404","Not here"); } exit; tnx miha    
>
>
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