[asterisk-users] bad sound quality after Redirect

[Atis Lezdins]

On 1/16/08, Franz Schwartau <franz at electromail.org> wrote:
> 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? :-)

Asterisk version?

First of all - i would try to identify problem, by redirecting to two
Dial's to separate SIP phones. That would tell if it's Redirect or
AppConference problem. Additionally you can try call that's not
Originate'd from manager.. Btw, why not using app_meetme, bundled with
Asterisk? Some time ago i was working on similar solution - sending
existing call to meetme and adding another Playback() to both calls,
however i currently don't have working version of this. You can try my
sample scenario from http://bugs.digium.com/view.php?id=10636


Regards,
Atis

-- 
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835