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@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@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@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@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
> > >> >

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