RES: RES: [Asterisk-Users] 403 Forbidden
Mireia Munoz de jesus
Mireia.Munoz-de-jesus at insa-lyon.fr
Fri Mar 12 06:50:37 MST 2004
The codecs are:
SIP Phone:
choice 1: PCMU
choice 2: PCMA
choice 3: G723
choice 4: G729
choice 5: G726-32
choice 6: G728
Asterisk:
in sip.conf
1: ulaw
2: alaw
in oh323.conf
1: G711U
Gateway:
preference 1: G711U
preference 2: ....
.
.
.
preference 8: G711A
That's good? Can you see where's the problem?
Thanks a lot for all your help.
Best Regards,
Mireia
Quoting Vinicius Viana <vinicius at telenova.net>:
> 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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