[asterisk-users] Possible bug in app_meetme.c

Richard Kenner kenner at gnat.com
Sat Oct 17 21:14:28 CDT 2009


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.



More information about the asterisk-users mailing list