Hi, On 5/5/05, Niksa Baldun <[EMAIL PROTECTED]> wrote: > 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.
I tried to put prefixes in h323.conf in following way, however it didn't work: [30598272] type=h323 prefix=1100001 prefix=1100005 prefix=1100006 prefix=1100007 context=home ;e164=1100007 -------------------------------------------------------------------------- Am I doing something wrong? Does somebody have configuration samples? thanks, Ganbold > > 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 > >[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
