RES: RES: [Asterisk-Users] 403 Forbidden
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 H245Received capability set, is accepted 46:01.165 H245:816f650 H245TerminalCapabilitySet already in progress: outSeq=1 46:01.165 H245:816f650 H245Sending PDU: response terminalCapabilitySetAck 46:01.166 H245:816f650 H323 InternalEstablishedConnectionCheck: connectionState=Await ingSignalConnect fastStartState=FastStartDisabled 46:01.167 H225 Caller:8141218 H225Set protocol version to 4 46:01.167 H225 Caller:8141218 H323Clearing connection ip$localhost/7705 reason=EndedByQ931C ause 46:01.167 H225 Caller:8141218 H323Call end reason for ip$localhost/7705 set to EndedByQ931C ause 46:01.167 H225 Caller:8141218 H225Sending release complete PDU: callRef=7705 46:01.170 H225 Caller:8141218 H245Sending PDU: command endSessionCommand 46:01.170 H225 Caller:8141218 H225Sending PDU: releaseComplete 46:01.171 H323 Cleaner H323Cleaning 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/listin
Re: RES: RES: [Asterisk-Users] 403 Forbidden
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 H245Received capability set, is > accepted > 46:01.165 H245:816f650 H245TerminalCapabilitySet > already in > progress: outSeq=1 > 46:01.165 H245:816f650 H245Sending PDU: response > terminalCapabilitySetAck > 46:01.166 H245:816f650 H323 > InternalEstablishedConnectionCheck: connectionState=Await > ingSignalConnect fastStartState=FastStartDisabled > 46:01.167 H225 Caller:8141218 H225Set protocol version to 4 > 46:01.167 H225 Caller:8141218 H323Clearing connection > ip$localhost/7705 reason=EndedByQ931C > ause > 46:01.167 H225 Caller:8141218 H323Call end reason for > ip$localhost/7705 set to EndedByQ931C > ause > 46:01.167 H225 Caller:8141218 H225Sending release complete > PDU: > callRef=7705 > 46:01.170 H225 Caller:8141218 H245Sending PDU: command > endSessionCommand > 46:01.170 H225 Caller:8141218 H225Sending PDU: releaseComplete > 46:01.171 H323 Cleaner H323Cleaning 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 be
Re: RES: RES: [Asterisk-Users] 403 Forbidden
Mireia Munoz de jesus wrote: 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 Try with one codec first (say G711A) in both SIP/H.323 channels. That's good? Can you see where's the problem? Thanks a lot for all your help. Best Regards, Mireia Michael. ___ 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
