Try setting canreinvite = no in sip.conf or the database (where you have sipuser setting).
Regards, Sanjay Rajdev ----- Original Message ----- From: "kalle odenthal" <[EMAIL PROTECTED]> To: asterisk-users@lists.digium.com Sent: Friday, March 30, 2007 5:52:47 AM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] SIP RTP Tunnel Hello, is it possible to rout ALL RTP Data over Asterisk, like SIP1 <---RTP---> Asterisk <---RTP---> SIP2 I know it seems quite useless. But I want to simulate a IAX -> SIP connection and have no Phonecard installed on my computer ;) Thanx, Kalle _______________________________________________ --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 _______________________________________________ --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
