> http://bugs.digium.com/bug_view_page.php?bug_id=0001589 > > 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.
Can't say that I've noticed any at all, but then most of our calls are tdm04b-fxo <--> 7960's with very little bridging. Since I recall you mentioning your use of 7960's previously, could the blip be related to the cisco issue/problem associated with slow startup and/or their v6.x code that drops rtp packets with inconsistence timestamps, combined with the usleep parameter? Rich _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
