Assuming your h.323 phones are registered with gnugk, you need to instruct gnugk to forward certain numbers to Asterisk. In OH323 (which I am using) you would need to add something like:
[register] gwprefix=0 gwprefix=1 etc. In h323.conf, I believe you have to add prefix=xxx in your endpoint definition. Bear in mind though that H.323 support in Asterisk is rather inadequate (only basic telephony functions are available). Niksa Baldun Ganbold Tsagaankhuu wrote: >Hi, > >I'm trying to configure asterisk to work with gnugk-2.0.8. Something like: > >SIP phones -> ASTERISK -> GNUGK ->Cisco GW -> PSTN > | > h323 phones > > >Following is h323.conf: >-------------------------------------------------------------------------------- >[general] >port = 1720 >bindaddr = 0.0.0.0 > >disallow=all >allow=g729 >gatekeeper = x.x.x.x >secret = 1234 >AllowGKRouted = yes >noFastStart = yes >noH245Tunneling = yes >noSilenceSuppression = yes > >[30598272] >type=h323 >prefix=1100001,1100005,1100006,1100007 >context=home >;e164=1100007 > >[1100005] >type=user >context=home >incominglimit=4 >-------------------------------------------------------------------- >sip.conf > >[general] >port=5060 ; Port to bind to >bindaddr=0.0.0.0 ; Address to bind SIP channel to >context=home ; Default context for incoming calls >musicclass=default >;videosupport=yes >allow=g729 >allow=g723 > >;externip = 202.179.0.164 >;localnet=192.168.0.0/255.255.0.0 > > >[1100001] >type=friend >username=1100001 >;secret=1111 >host=dynamic >nat=yes >defaultip=192.168.0.11 >context=home >canreinvite=no >callerid=1100001 >[EMAIL PROTECTED] > >[1100002] >type=friend >username=1100002 >;secret=2222 >nat=yes >host=dynamic >context=home >canreinvite=no >callerid=1100002 >[EMAIL PROTECTED] > >[1100005] >type=friend >username=1100005 >;secret=1234 >defaultip=192.168.0.62 >nat=yes >host=dynamic >context=home >canreinvite=no >callerid=1100005 >[EMAIL PROTECTED] > >[1100006] >type=friend >username=1100006 >;secret=4321 >host=dynamic >context=home >canreinvite=no >callerid=1100006 >[EMAIL PROTECTED] >-------------------------------------------------------------------------------- > >As in above configuration I'm registering Asterisk as an endpoint to gnugk. >It is working and I can make calls from SIP phones to PSTN. >However my question is, how can I call from h323 endpoints to SIP >phones or vice versa in above case? >Is it possible? I'm afraid, it can't since asterisk is itself an one >endpoint to gnugk. >If possible how can I make it work? > >If not, is it possible to register or make each SIP phones to be known to >gnugk? >How can I accomplish that? Ideally this solution could be the best. > >It would be very helpful if somebody can show me the config samples. > >thanks in advance, > >Ganbold >_______________________________________________ >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 > > > > _______________________________________________ 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