[asterisk-dev] how to enable 16Khz audio conference in Asterisk+ConfBridge

[Weikai Xie]

Hi, folks,
    I learned from document the new ConfBridge application can do
audio mix in 16khz mode. However, I could not successfully  achieve
this in my experiement. Here is my configuration:

1. Asterisk 1.6.2.7 server running on CentOS 5.3
2. I added following directives in extensions.conf
    extern => 500,1,Answer()
    extern => 500,2,ConfBridge(100)
3. I intended to use 16Khz Speex as the audio codec in the conference.
So I add following lines to the default section in sip.conf as well
    disallow=all
    allow=speex

3. I use two SIP UA to call into the extension 500. The specific UA I
used is the  PJSUA command line based SIP UA, which give the 16Khz
mode Speex the highest preference.

The call can be established. However, the dump of the PJSUA shows the
actual codec used is the 8Khz Speex.  I guess somehow the Asterisk
refused to use 16Khz speex during the format negotiation.

My questions is what I should do to instruct Asterisk and Confbridge
to use 16Khz end to end in such scenario? Thanks.



Regards.

Weikai Xie