David Backeberg wrote: > From a quick glance at your patch, I would say that it probably tries > to address the audio quality problems I and others were experiencing.
No, it's fixing a much more serious issue. As I sent to this list twice, when I have a conference between Dahdi ports and SIP phones, the people on the SIP phones were never heard. I had the 'o' option specified. The code, as written, says "for non-Dahdi (e.g., SIP phones), send to the channel if talking AND not optimizing talkers". It seems to me that it should send to the challel if talking OR not optimizing talkers. _______________________________________________ -- 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
