[Asterisk-Users] SIP rtp port forcing

[Kannaiyan Natesan]

check   rtp.conf

-Kannaiyan
  ----- Original Message ----- 
  From: boris.vincent at mindspeed.com 
  To: asterisk-users at lists.digium.com 
  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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