Also set canreinvite=no between Asterisk and the provider.
[EMAIL PROTECTED] wrote:
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
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