search for: 1f842ecf

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2004 Sep 06
3
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: