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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nome de Mireia Munoz de jesus Enviada em: quarta-feira, 10 de março de 2004 16:46 Para: [EMAIL PROTECTED]; Martin Mielke Cc: [EMAIL PROTECTED] 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 <[EMAIL PROTECTED]>: > 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 > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.602 / Virus Database: 383 - Release Date: 01/03/2004 --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.602 / Virus Database: 383 - Release Date: 01/03/2004 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users