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
