[asterisk-users] SIP RTP Tunnel

[Sanjay Rajdev]

Try setting canreinvite = no in sip.conf or the database (where you have sipuser setting).

Regards,
Sanjay Rajdev

----- Original Message -----
From: "kalle odenthal" <kalle.odenthal at genion.de>
To: asterisk-users at 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




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