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Has anyone else heard an audible blip, break or
garble between answer and the native bridge attempt using sip?
If I change the usleep(500000); to usleep(5000); in
rtp.c the proble totally goes away... even the note above it says it needs
to be fixed.
bkw
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- Re: [Asterisk-Users] 500ms usleep in rtp.c ? brian k. west
