RES: RES: [Asterisk-Users] 403 Forbidden

Vinicius Viana vinicius at telenova.net
Thu Mar 11 09:14:21 MST 2004 

The call end reason "EndedByQ931Cause" is used by the OpenH323 stack when it
doesn't know the real cause.
Try to see if the codecs in the gateway are compatible with the codecs in
asterisk.
What are the codecs you are using in SIP Phones, in Asterisk and in the
gateway?

Regards,

Vinicius



-----Mensagem original-----
De: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]Em nome de Mireia Munoz de
jesus
Enviada em: quinta-feira, 11 de março de 2004 11:37
Para: asterisk-users at lists.digium.com; Vinicius Viana
Assunto: Re: RES: [Asterisk-Users] 403 Forbidden


Hi, thanks a lot for your answer. When I call from SIP phone to analogic
found I
get this log file:

(I only show, when there's the disconnection)
....
46:01.165                 H245:816f650 H245    Received capability set, is
accepted
 46:01.165                 H245:816f650 H245    TerminalCapabilitySet
already in
progress: outSeq=1
 46:01.165                 H245:816f650 H245    Sending PDU: response
terminalCapabilitySetAck
 46:01.166                 H245:816f650 H323
InternalEstablishedConnectionCheck: connectionState=Await
ingSignalConnect fastStartState=FastStartDisabled
 46:01.167          H225 Caller:8141218 H225    Set protocol version to 4
 46:01.167          H225 Caller:8141218 H323    Clearing connection
ip$localhost/7705 reason=EndedByQ931C
ause
 46:01.167          H225 Caller:8141218 H323    Call end reason for
ip$localhost/7705 set to EndedByQ931C
ause
 46:01.167          H225 Caller:8141218 H225    Sending release complete
PDU:
callRef=7705
 46:01.170          H225 Caller:8141218 H245    Sending PDU: command
endSessionCommand
 46:01.170          H225 Caller:8141218 H225    Sending PDU: releaseComplete
 46:01.171                 H323 Cleaner H323    Cleaning up connections

I suppose, from what you have told me in your mail, that the problem is in
my
gateway.... so, have you any idea what can be the exact problem and how to
solve it?

Thanks a lot for you answer.

Best Regards,

Mireia

Quoting Vinicius Viana <vinicius at telenova.net>:

> I believe your gatekeeper or your gateway is refusing the call. This can
be
> a authorization problem in the gatekeeper or codec problem in the gateway.
>
> You need to see where your call is failing. Try to do the following:
>
> 1 - Turn on the oh323 trace in the oh323.conf file adding these lines to
> your configuration:
> wrapLibTraceLevel=3
> libTraceLevel=3
> libTraceFile=/var/log/asterisk/oh323.log
>
> 2 - Make a call from your SIP Phone to your PBX
>
> 3 - Look into the /var/log/asterisk/oh323.log and verify if the call is
> failing in the Admission Request or in the Setup message.
>
> 4 - If it fails in the Admission Request (you will see a Admission Reject
> into the log) the problem is in the configuration of your gatekeeper.
> 5 - If it fails in the Setup message (you will see a Release Complete into
> the log) the problem is in the configuration of your gateway
>
> Other thing you can see is if your asterisk box is registered with your
> gatekeeper.
>
> With the information you supplied this is what I remember you can check to
> see what is wrong.
>
> Regards,
>
> Vinicius
>
> -----Mensagem original-----
> De: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com]Em nome de Mireia Munoz de
> jesus
> Enviada em: quarta-feira, 10 de março de 2004 16:46
> Para: asterisk-users at lists.digium.com; Martin Mielke
> Cc: asterisk-users at lists.digium.com
> Assunto: Re: [Asterisk-Users] 403 Forbidden
>
>
> Hi,
>
> Thanks for your answer, but my asterisk is working as a H.323 - SIP
gateway
> and
> calls between SIP clients (phone and soft clients) are working all right.
> The
> only problem I have, is like I have said in my mail is between sip phones
> and
> PBX.
>
> Best Regards,
>
> Mireia
>
> PS: Someone have other ideas?
>
>
> Quoting Martin Mielke <martin.mielke at thales-is.com>:
>
> > Hi Mieria,
> >
> > Mireia Munoz de jesus wrote:
> >
> > >Hi!
> > >
> > >When I try to call from a SIP phone to a PBX phone I get this error:
> > >
> > >chan_oh323.c [1004] Couldn`t call 483377839
> > >
> > >and if I get the messages from SIP debug, I have a 403 message. The
> > >configuration of my system is:
> > >
> > >SIP Phone ---- ASterisk ---- Gatekeeper ----- Gateway ----- PBX -----
> Phone
> > >
> > >Have someone any idea of what is going on?. It will be very nice if
> someone
> > >helps... it`s been more than a week that I can`t solve this problem.
> > >
> > >Best Regards,
> > >
> > >Mireia
> > >
> >
> > Could it be that  you are using a *SIP* phone? Although you can add
> > H.323 to Asteriskm, SIP and H.323 are different protocols...
> >
> >
> > HTH,
> >
> > Martin
> >
> >
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>
>
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