check rtp.conf
-Kannaiyan
- Original Message -
From:
[EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, September 06, 2004 6:15
PM
Subject: [Asterisk-Users] SIP rtp port
forcing
When a SIP call starts (INVITE
/ 200 OK), asterisk seems to create a
You can only restrict the range of ports used, in rtp.conf.
I suppose restricting it to 2 ports starting on even number might do it,
but if you're not using SIP on one end, how are you going to start a call?
You need to have at least rudimentary call control for SIP invite and SDP
exchange, and
