On Sat, Oct 17, 2009 at 2:26 PM, Richard Kenner <ken...@gnat.com> wrote: > Is this patch correct? The "&&" doesn't make logical sense to me. I think > it should be "||" and making this change fixes the problem I have with SIP > phones in MeetMe conferences. If it's correct, is there someplace more > formal that I should submit it to?
So now I actually read through your patch. Please read my last post about getting up to speed with how the history of talker optimization with MeetMe() in 1.6 has gone. Having said that, I think you may be using talker optimization and that this was causing the same voice cut out problem I was having. What version are you running? Does that version support disabling talker optimization? Have you tried disabling talker optimization? I appreciate your work on this, but I fear your issues may have already been solved a rather long time ago. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
