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 given that you now have SDP exchange you should be able to accept any port presented by asterisk.
[EMAIL PROTECTED] wrote:
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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