Hola Sanjay, this works pretty well in one direction. The Sip User who is registered at the Asterisk. But the Sip user who calls from sipXYZ.com still sends it data diretly to sip user 1.
Any idea? Thanx!! -----Original Message----- From: Sanjay Rajdev [mailto:[EMAIL PROTECTED] Sent: Donnerstag, 29. März 2007 18:27 To: kalle odenthal; Asterisk Users Mailing List - Non-Commercial Discussion Cc: [email protected] Subject: Re: [asterisk-users] SIP RTP Tunnel 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: [email protected] 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
