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Hello<br>
I also have a similar problem. The dialog module doesn't detect the BYE
message.<br>
I'm using ver 1.1.1.<br>
My configuration is as follow: 2 Wifi SIP phones (BCM) connected to the
same Access Point and the OpenSER runs on a PC.<br>
Attached the debug log, ethereal sniffing on the <b>Wire</b> LAN and
my config file.<br>
For both ACK and BYE message, the dialog module prints the error <br>
DEBUG:dialog:dlg_onroute: Route param 'did' not found<br>
Did you find a solution?<br>
<br>
If you want to check the attached files:<br>
Caller: 192.168.13.166<br>
Callee: 192.168.13.101<br>
SIP Proxy: 192.168.13.86<br>
<br>
Regards,<br>
Michel.<br>
<br>
<br>
Bogdan-Andrei Iancu wrote:
<blockquote cite="mid45DEF233.5030802@voice-system.ro" type="cite">Hi
Andy,
<br>
<br>
in client config, you need to add "[routes]" for ACK and BYE messages
(take a look at the cfg I sent you)
<br>
<br>
regards,
<br>
bogdan
<br>
<br>
Andy Pyles wrote:
<br>
<blockquote type="cite">I Just re-read the docs on loose_route(). So
please disregard this
<br>
question. ( only processed if Route: header is present. Which isn't
<br>
present because Record-route: header isn't being sent to caller )
<br>
<br>
So, I'm still trying to figure out why record-route: header is not
<br>
being sent to caller.
<br>
<br>
<br>
On 2/22/07, Andy Pyles <a class="moz-txt-link-rfc2396E" href="mailto:andy.pyles@gmail.com"><andy.pyles@gmail.com></a> wrote:
<br>
<blockquote type="cite">Hi Bogdan,
<br>
<br>
After running additional debugs, for some reason the call to
<br>
loose_route() is failing.
<br>
<br>
if (loose_route()) {
<br>
# mark routing logic in request
<br>
xlog("L_INFO", "loose_route() succeeded\n ");
<br>
route(1);
<br>
} else{
<br>
xlog("L_INFO", "loose_route()failed - M=$rm RURI=$ru F=$fu
<br>
T=$tu IP=$si ID=$ci\n");
<br>
};
<br>
<br>
<br>
Any ideas why this could be occuring?
<br>
<br>
<br>
On 2/22/07, Andy Pyles <a class="moz-txt-link-rfc2396E" href="mailto:andy.pyles@gmail.com"><andy.pyles@gmail.com></a> wrote:
<br>
> HI Bogdan,
<br>
>
<br>
> I'm already using an almsot identical version of uas.xml and
uac.xml (
<br>
> yes rrs=true) is being used. However in your version the uas.xml
<br>
> doesn't have rrs="true" after initial invite which I think is
needed.
<br>
> See as you can see below, setting rrs="true" for uac will only
work if
<br>
> it receives a Record-Route header in the 200OK which it's not.
<br>
>
<br>
> In this case, ALL messages from openser to sipp uac do not contain
the
<br>
> Record-route header. So I don't think it's a sipp problem, but an
<br>
> openser configuration problem. I've tried using other devices for
a
<br>
> uac, such as x-lite but the same problem.
<br>
>
<br>
> Andy
<br>
>
<br>
> On 2/22/07, Bogdan-Andrei Iancu <a class="moz-txt-link-rfc2396E" href="mailto:bogdan@voice-system.ro"><bogdan@voice-system.ro></a>
wrote:
<br>
> > Hi Andy,
<br>
> >
<br>
> > so it's about sipp :D - I remember I had some hard times to
make it work
<br>
> > with record Route.
<br>
> >
<br>
> > take a look at the attached files, they might help you.
<br>
> >
<br>
> > regards,
<br>
> > bogdan
<br>
> >
<br>
> > Andy Pyles wrote:
<br>
> > > HI Bogdan,
<br>
> > >
<br>
> > > thanks for your reply.
<br>
> > > yes you are correct. The Bye doesn't have the Route
header.
<br>
> > > It appears the the 200 OK sent to the caller doesn't
contain a
<br>
> > > Record-route header.
<br>
> > > Messages between openser and callee contain record-route
information,
<br>
> > > but messages between caller and openser do not.
<br>
> > > Is there a way to enable that?
<br>
> > >
<br>
> > > Here's more detail:
<br>
> > > 192.168.0.101 = Caller (sipp)
<br>
> > > 1.2.3.4 = openser
<br>
> > > 4.3.2.1 = callee ( sipp)
<br>
> > >
<br>
> > >
<br>
> > > 1.) 192.168.0.101 -> 1.2.3.4 SIP/SDP Request:
INVITE
<br>
> > > <a class="moz-txt-link-abbreviated" href="mailto:sip:service@1.2.3.4:5060">sip:service@1.2.3.4:5060</a>, with session description
<br>
> > > 2.) 1.2.3.4 -> 192.168.0.101 SIP Status: 100 Giving
a try
<br>
> > > 3.) 1.2.3.4 -> 4.3.2.1 SIP/SDP Request: INVITE
<br>
> > > <a class="moz-txt-link-abbreviated" href="mailto:sip:service@4.3.2.1:5060">sip:service@4.3.2.1:5060</a>, with session description
<br>
> > > 4.) 4.3.2.1 -> 1.2.3.4 SIP Status: 180
Ringing
<br>
> > > 5.) 4.3.2.1 -> 1.2.3.4 SIP/SDP Status: 200
OK, with session
<br>
> > > description
<br>
> > > 6.) 1.2.3.4 -> 192.168.0.101 SIP Status: 180
Ringing
<br>
> > > 7.) 1.2.3.4 -> 192.168.0.101 SIP/SDP Status: 200
OK, with session
<br>
> > > description
<br>
> > > 8.) 192.168.0.101 -> 1.2.3.4 SIP Request:
ACK
<br>
> > > <a class="moz-txt-link-abbreviated" href="mailto:sip:service@1.2.3.4:5060">sip:service@1.2.3.4:5060</a>
<br>
> > > 9.) 1.2.3.4 -> 4.3.2.1 SIP Request: ACK
<a class="moz-txt-link-abbreviated" href="mailto:sip:service@4.3.2.1:5060">sip:service@4.3.2.1:5060</a>
<br>
> > > 10.) 192.168.0.101 -> 1.2.3.4 SIP Request: BYE
<br>
> > > <a class="moz-txt-link-abbreviated" href="mailto:sip:service@1.2.3.4:5060">sip:service@1.2.3.4:5060</a>
<br>
> > > 11.) 1.2.3.4 -> 4.3.2.1 SIP Request: BYE
<a class="moz-txt-link-abbreviated" href="mailto:sip:service@4.3.2.1:5060">sip:service@4.3.2.1:5060</a>
<br>
> > > 12.) 4.3.2.1 -> 1.2.3.4 SIP Status: 200 OK
<br>
> > > 13.) 1.2.3.4 -> 192.168.0.101 SIP Status: 200 OK
<br>
> > >
<br>
> > > ---
<br>
> > > Packets 6,7 and following contain no Record-route
information.
<br>
> > > The other weird thing is that openser is passing on the
Route: header
<br>
> > > it recevied from callee to the caller.
<br>
> > >
<br>
> > >
<br>
> > > Please see attached for complete ngrep output.
<br>
> > >
<br>
> > >
<br>
> > > On 2/21/07, Bogdan-Andrei Iancu
<a class="moz-txt-link-rfc2396E" href="mailto:bogdan@voice-system.ro"><bogdan@voice-system.ro></a> wrote:
<br>
> > >> Hi Andy,
<br>
> > >>
<br>
> > >> could you check on the net if the BYE contain the
Route hdr added to
<br>
> > >> INVITE as Record-Route? I have some doubts on this
as I see:
<br>
> > >> 0(966) find_first_route: No Route headers found
<br>
> > >> 0(966) loose_route: There is no Route HF
<br>
> > >>
<br>
> > >> and if the BYE is not identified, the dialog is not
closed.
<br>
> > >>
<br>
> > >> regards,
<br>
> > >> bogdan
<br>
> > >>
<br>
> > >> Andy Pyles wrote:
<br>
> > >> > Hello,
<br>
> > >> >
<br>
> > >> > I have a question on how to configure the
dialog module ( 1.2.x from
<br>
> > >> > cvs yesterday ).
<br>
> > >> >
<br>
> > >> > With my config, ( attached) I can make calls
and have verified that
<br>
> > >> > the acc module is working correctly.
<br>
> > >> >
<br>
> > >> > My question is, when I enable the dialog
module, I can see that it is
<br>
> > >> > incrementing call count correctly, but when a
bye is received, the
<br>
> > >> > dialog:active_dialogs statistic is never
decremented.
<br>
> > >> >
<br>
> > >> > In the debug level 9 logs, ( also attached) I
see this error after the
<br>
> > >> > 200OK is sent to the bye:
<br>
> > >> >
<br>
> > >> > 1(969) DBUG:dialog:unref_dlg: unref dlg
0xa7ce5a98 with 1
<br>
> > >> (delete=0)-> 1
<br>
> > >> >
<br>
> > >> > Is this a case of one of the timers being set
too short? by the way
<br>
> > >> > using a variable call length from well under
a second ( using sipp )
<br>
> > >> > to 20 second call doesnt' seem to make a
difference .
<br>
> > >> >
<br>
> > >> >
<br>
> > >> > Thanks,
<br>
> > >> > Andy
<br>
> > >> >
<br>
</blockquote>
</blockquote>
<br>
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