[Users] dialog module configuration question
Bogdan-Andrei Iancu
bogdan at voice-system.ro
Sun Feb 25 16:05:32 CET 2007
Hi Michel,
looking at the net capture, it seams that the UAS device (User-Agent:
WLAN660-S VoIP PHONE) does not correctly mirror the RR header - it is
removing the hdr parameters, mirroring only the URI, which is bogus.
regards,
bogdan
Michel Bensoussan wrote:
> Hello
> I also have a similar problem. The dialog module doesn't detect the
> BYE message.
> I'm using ver 1.1.1.
> My configuration is as follow: 2 Wifi SIP phones (BCM) connected to
> the same Access Point and the OpenSER runs on a PC.
> Attached the debug log, ethereal sniffing on the *Wire* LAN and my
> config file.
> For both ACK and BYE message, the dialog module prints the error
> DEBUG:dialog:dlg_onroute: Route param 'did' not found
> Did you find a solution?
>
> If you want to check the attached files:
> Caller: 192.168.13.166
> Callee: 192.168.13.101
> SIP Proxy: 192.168.13.86
>
> Regards,
> Michel.
>
>
> Bogdan-Andrei Iancu 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 at 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 at 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 at 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 at 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 at 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 at 1.2.3.4:5060
>>>> > > > 9.) 1.2.3.4 -> 4.3.2.1 SIP Request: ACK
>>>> sip:service at 4.3.2.1:5060
>>>> > > > 10.) 192.168.0.101 -> 1.2.3.4 SIP Request: BYE
>>>> > > > sip:service at 1.2.3.4:5060
>>>> > > > 11.) 1.2.3.4 -> 4.3.2.1 SIP Request: BYE
>>>> sip:service at 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 at 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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