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 <[EMAIL PROTECTED]>: > 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: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] nome de Mireia Munoz de > jesus > Enviada em: quinta-feira, 11 de março de 2004 11:37 > Para: [EMAIL PROTECTED]; 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 <[EMAIL PROTECTED]>: > > > 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 > > > > > _______________________________________________ > 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 > _______________________________________________ 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