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 >