[asterisk-users] bad sound quality after Redirect

[Franz Schwartau]

Hi!

I'm building an application which allows to dial via the Asterisk 
Manager Interface using the originate command. There should be an 
optional conferencing feature.

The manager commands are basically:

---------------------------------
action: login
username: sdjklgdsjg
secret: xxx
events: on

action: originate
callerid: 3847438609
priority: 1
exten: 4068439865
async: 1
context: out
channel: SIP/sip-gate/0394839405
---------------------------------

Then talk to each other for a while...

---------------------------------
action: redirect
priority: 1
exten: 1234
context: conference
channel: SIP/sip-gate-0868b000
extrachannel: SIP/sip-gate-086a5000

action: logoff
---------------------------------

This approach works but results in a bad sound quality after the 
redirect. The sound seems to be scrambled. Before redirecting the sound 
quality is quite well, of course. All extensions are called via SIP with 
the same codec, so no transcoding should occur.

The application used for the conference room is AppConference from 
http://sourceforge.net/projects/appconference/. But even with a simple 
destination application (e. g. PlayTones or Playback) the sound quality 
is as bad as with AppConference. So it doesn't seem to be a problem with 
AppConference itself.

The bad sound quality arises only if the ExtraChannel parameter is given 
to Redirect. Without ExtraChannel the sound quality is still fine. But 
the second channel is hungup then of course, which is not intended.

Has anyone any ideas how to solve this problem? :-)

	Best regards Franz