[Asterisk-Users] sip to oh323 converter converts sip uri to h.323 number and not h.323 url
Hello, i have set up an asterisk sip to h.323 convertor, it is working OK. The only problem i have is this : For example when my identity is [EMAIL PROTECTED] , and i call a sip number from a sip phone, the called party sees my identity (caller identity) as [EMAIL PROTECTED], which is the way it has to be. But when i call from the same phone with the same identity a h.323 endpoint (asterisk converts), the h.323 endpoints sees my identyty as '12345'. So asterisk is deleting everything after the @ (included). How can i make that the oh323/asterisk sends the whole SIP URIas caller identity to the H.323 network? Thankk you Oliver ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip to oh323 converter converts sip uri to h.323 number and not h.323 url
Hello, i have set up an asterisk sip to h.323 convertor, it is working OK. The only problem i have is this : For example when my identity is [EMAIL PROTECTED] , and i call a sip number from a sip phone, the called party sees my identity (caller identity) as [EMAIL PROTECTED], which is the way it has to be. But when i call from the same phone with the same identity a h.323 endpoint (asterisk converts), the h.323 endpoints sees my identyty as '12345'. So asterisk is deleting everything after the @ (included). How can i make that the oh323/asterisk sends the whole SIP URI as caller identity to the H.323 network? Thankk you Oliver ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
