Hi Bogdan, correct. but on client config "[routes]" ( for sipp) will only work IF the client receives a Record-route. Since I'm not, it doesn't help me. Am I missing something? Andy On 2/23/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: > Hi Andy, > > in client config, you need to add "[routes]" for ACK and BYE messages > (take a look at the cfg I sent you) > > regards, > bogdan > > Andy Pyles wrote: > > I Just re-read the docs on loose_route(). So please disregard this > > question. ( only processed if Route: header is present. Which isn't > > present because Record-route: header isn't being sent to caller ) > > > > So, I'm still trying to figure out why record-route: header is not > > being sent to caller. > > > > > > On 2/22/07, Andy Pyles <andy.pyles(a)gmail.com> wrote: > >> Hi Bogdan, > >> > >> After running additional debugs, for some reason the call to > >> loose_route() is failing. > >> > >> if (loose_route()) { > >> # mark routing logic in request > >> xlog("L_INFO", "loose_route() succeeded\n "); > >> route(1); > >> } else{ > >> xlog("L_INFO", "loose_route()failed - M=$rm RURI=$ru F=$fu > >> T=$tu IP=$si ID=$ci\n"); > >> }; > >> > >> > >> Any ideas why this could be occuring? > >> > >> > >> On 2/22/07, Andy Pyles <andy.pyles(a)gmail.com> wrote: > >> > HI Bogdan, > >> > > >> > I'm already using an almsot identical version of uas.xml and uac.xml ( > >> > yes rrs=true) is being used. However in your version the uas.xml > >> > doesn't have rrs="true" after initial invite which I think is needed. > >> > See as you can see below, setting rrs="true" for uac will only work if > >> > it receives a Record-Route header in the 200OK which it's not. > >> > > >> > In this case, ALL messages from openser to sipp uac do not contain the > >> > Record-route header. So I don't think it's a sipp problem, but an > >> > openser configuration problem. I've tried using other devices for a > >> > uac, such as x-lite but the same problem. > >> > > >> > Andy > >> > > >> > On 2/22/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: > >> > > Hi Andy, > >> > > > >> > > so it's about sipp :D - I remember I had some hard times to make > >> it work > >> > > with record Route. > >> > > > >> > > take a look at the attached files, they might help you. > >> > > > >> > > regards, > >> > > bogdan > >> > > > >> > > Andy Pyles wrote: > >> > > > HI Bogdan, > >> > > > > >> > > > thanks for your reply. > >> > > > yes you are correct. The Bye doesn't have the Route header. > >> > > > It appears the the 200 OK sent to the caller doesn't contain a > >> > > > Record-route header. > >> > > > Messages between openser and callee contain record-route > >> information, > >> > > > but messages between caller and openser do not. > >> > > > Is there a way to enable that? > >> > > > > >> > > > Here's more detail: > >> > > > 192.168.0.101 = Caller (sipp) > >> > > > 1.2.3.4 = openser > >> > > > 4.3.2.1 = callee ( sipp) > >> > > > > >> > > > > >> > > > 1.) 192.168.0.101 -> 1.2.3.4 SIP/SDP Request: INVITE > >> > > > sip:service@1.2.3.4:5060, with session description > >> > > > 2.) 1.2.3.4 -> 192.168.0.101 SIP Status: 100 Giving a try > >> > > > 3.) 1.2.3.4 -> 4.3.2.1 SIP/SDP Request: INVITE > >> > > > sip:service@4.3.2.1:5060, with session description > >> > > > 4.) 4.3.2.1 -> 1.2.3.4 SIP Status: 180 Ringing > >> > > > 5.) 4.3.2.1 -> 1.2.3.4 SIP/SDP Status: 200 OK, with > >> session > >> > > > description > >> > > > 6.) 1.2.3.4 -> 192.168.0.101 SIP Status: 180 Ringing > >> > > > 7.) 1.2.3.4 -> 192.168.0.101 SIP/SDP Status: 200 OK, with > >> session > >> > > > description > >> > > > 8.) 192.168.0.101 -> 1.2.3.4 SIP Request: ACK > >> > > > sip:service@1.2.3.4:5060 > >> > > > 9.) 1.2.3.4 -> 4.3.2.1 SIP Request: ACK > >> sip:service@4.3.2.1:5060 > >> > > > 10.) 192.168.0.101 -> 1.2.3.4 SIP Request: BYE > >> > > > sip:service@1.2.3.4:5060 > >> > > > 11.) 1.2.3.4 -> 4.3.2.1 SIP Request: BYE > >> sip:service@4.3.2.1:5060 > >> > > > 12.) 4.3.2.1 -> 1.2.3.4 SIP Status: 200 OK > >> > > > 13.) 1.2.3.4 -> 192.168.0.101 SIP Status: 200 OK > >> > > > > >> > > > --- > >> > > > Packets 6,7 and following contain no Record-route information. > >> > > > The other weird thing is that openser is passing on the Route: > >> header > >> > > > it recevied from callee to the caller. > >> > > > > >> > > > > >> > > > Please see attached for complete ngrep output. > >> > > > > >> > > > > >> > > > On 2/21/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: > >> > > >> Hi Andy, > >> > > >> > >> > > >> could you check on the net if the BYE contain the Route hdr > >> added to > >> > > >> INVITE as Record-Route? I have some doubts on this as I see: > >> > > >> 0(966) find_first_route: No Route headers found > >> > > >> 0(966) loose_route: There is no Route HF > >> > > >> > >> > > >> and if the BYE is not identified, the dialog is not closed. > >> > > >> > >> > > >> regards, > >> > > >> bogdan > >> > > >> > >> > > >> Andy Pyles wrote: > >> > > >> > Hello, > >> > > >> > > >> > > >> > I have a question on how to configure the dialog module ( > >> 1.2.x from > >> > > >> > cvs yesterday ). > >> > > >> > > >> > > >> > With my config, ( attached) I can make calls and have > >> verified that > >> > > >> > the acc module is working correctly. > >> > > >> > > >> > > >> > My question is, when I enable the dialog module, I can see > >> that it is > >> > > >> > incrementing call count correctly, but when a bye is > >> received, the > >> > > >> > dialog:active_dialogs statistic is never decremented. > >> > > >> > > >> > > >> > In the debug level 9 logs, ( also attached) I see this error > >> after the > >> > > >> > 200OK is sent to the bye: > >> > > >> > > >> > > >> > 1(969) DBUG:dialog:unref_dlg: unref dlg 0xa7ce5a98 with 1 > >> > > >> (delete=0)-> 1 > >> > > >> > > >> > > >> > Is this a case of one of the timers being set too short? by > >> the way > >> > > >> > using a variable call length from well under a second ( > >> using sipp ) > >> > > >> > to 20 second call doesnt' seem to make a difference . > >> > > >> > > >> > > >> > > >> > > >> > Thanks, > >> > > >> > Andy > >> > > >> > >