|
check rtp.conf
-Kannaiyan
----- Original Message -----
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 random port number for voice (rtp)
packets. Is it possible to force this port value (without using reinvite since
i am trying to use SIP against something else than sip)
thanks a lot in advance
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