[asterisk-users] asterisk 1.6.1.0 and dial plan changes

[David Backeberg]

On Sun, May 31, 2009 at 3:51 PM, sean darcy <seandarcy2 at gmail.com> wrote:
> David Backeberg wrote:
>>
>> You don't say the kind of call you're making, but if you're using
>> MeetMe() I have more advice regarding voice quality with conference
>> rooms.
>>
>
> I don't know about the OP, I'd sure appreciate any advice regarding
> voice quality with MeetMe(). When we have 2 -3 internal SIP lines, 2+
> internet SIP lines, and some PRI lines, we have a difficult time with
> quality.
>
> Any tips appreciated.

Sure. In addition to the things I mentioned, try jumping to the
1.6.1.* series. And be sure to NOT pass 'o' as an option to the
conference.

The 1.6.0. series had hard-coded talker optimization, which probably
makes things nice for very heavily loaded conferences, but for our
conferences was seeming to cause dropped voice packets that I assume
were mistaken for line noise. We were able to reliably produce lost
packets by making voice noises like breathing into the receiver, or
moaning at the right pitch. In addition to those problems, it would
clip the beginnings and endings of phrases. So if you were trying to
tell somebody your phone number, like 555 555 5555, with breaks
between each, you would have a very frustrating experience. You can
read about a lengthy discussion on making optimization optional rather
than mandatory at:

https://issues.asterisk.org/view.php?id=13801