I also have problems the same with you. I can find my asterisk registered on my GK's status port(7000). And I make a call from my XPro to SIP and SIP to Asterisk, then Asterisk calls to a H323 phone via GNUGK.
I can find the CDR message on GK's status monitor. But I the GK only first ACF(tail with false) and unconnected CDR on my GK. Do you solve your problem? Can you share me your asterisk config such as extensions.conf and oh323.conf, gatekeeper.ini for me to refer? Please. Best Regards Charles On Thu, 10 Mar 2005 23:33:53 -0800 (PST), Kamran Ahmad <[EMAIL PROTECTED]> wrote: > hello > > i am using my own gnugatekeeper as a gatekeeper for my > asterisk. asterisk is registering successfully with > Gnugatekeeper. but it is not transfering call to > gnugk. > > any one guide me who to do this > -------------------------------------------------- > SJPhone(sipSoftPhone using sip)->asterisk > asterisk(conversion from sip -> h.323) > asterisk(send h.323)->GnuGK > GnuGk->SoftPhone(h.323 OpenPhone) > ------------------------------------------------- > > on GnuGatekeeper side > gatekeeper.ini > ---------------------------------------- > [Gatekeeper::Main] > Fourtytwo=42 > TimeToLive=600 > > [RoutedMode] > GKRouted=1 > H245Routed=0 > CallSignalPort=1721 > > [RasSrv::PermanentEndpoints] > 192.168.0.203=xyz;123 > > [GkStatus::Auth] > rule=allow > > on asterisk > oh323.conf > --------------- > ; > ; Configuration file of OpenH323 channel driver > ; > > ;----------------------------------------- > ; General configuration options > ; (ports, jitter, GK, ...) > ;----------------------------------------- > [general] > listenAddress=192.168.0.203 > listenPort=1719 > connectPort=1719 > > tcpStart=10000 > tcpEnd=20000 > > udpStart=10000 > udpEnd=20000 > > fastStart=yes > > h245Tunnelling=no > > h245inSetup=no > > inBandDTMF=yes > > silenceSuppression=no > > jitterMin=20 > jitterMax=100 > > ipTos=none > tos=lowdelay > outboundMax=10 > inboundMax=10 > simultaneousMax=10 > > wrapLibTraceLevel=1 > libTraceLevel=1 > libTraceFile=stdout > > gatekeeper=192.168.0.153 > gatekeeperPassword=test1 > accountcode=test1 > gatekeeperTTL=600 > > userInputMode=TONE > > amaFlags=default > > context=default > > [xyz] > type=h323 > prefix=123 > context=default > > alias=1234 > context=default > ;----------------------------------------- > ; Specify and configure CODEC related > ; options > ;----------------------------------------- > [codecs] > codec=G711U > frames=20 > > extensions.conf > ------------------ > [default] > exten=>2000,1,Dial(SIP/${EXTEN}) > exten=>3000,1,Dial(SIP/${EXTEN}) > exten=>_123XXXX,1,Dial(SIP/${EXTEN}) > exten=>_321XXXX,1,Dial(OH323:h323/[EMAIL PROTECTED]:1719|30|r) > > sip.conf > ------------------ > [2000] > host=dynamic > type=friend > dtmfmode=INFO > canreinvite=no > > [3000] > host=dynamic > type=friend > dtmfmode=INFO > canreinvite=no > > __________________________________ > Do you Yahoo!? > Yahoo! Mail - now with 250MB free storage. Learn more. > http://info.mail.yahoo.com/mail_250 > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Best Regards Charles _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users