Thank you Salvatore, I have Wireshark ;) But my problem is not to SEE the packets. I'm trying to do a Handover between VoIP and GSM using an Asterisk Server. But to avoid the effort to implement the channel: Asterisk->public switched network->GSM->...radio...->GSM modem-> SmartPhone, I want to substitute the "GSM" Channel with SIP. This is valid for my work. The problem is if Asterisk DO tunnel all Packets through its bones when I switch between SIP and GSM, it connects two SIP participants direcly. You see my problem?
Saludos, Kalle You should get a packet capture and look at the SDP that is agreed to by both parties. It sounds like someone is not honoring it. -------------------------------------------------- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, March 29, 2007 9:35 PM To: [email protected] Subject: RE: RE: [asterisk-users] SIP RTP Tunnel 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 _______________________________________________ --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
