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franz at electromail.org
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 Posted: Wed Jan 16, 2008 5:18 am Post subject: [asterisk-users] bad sound quality after Redirect |
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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 |
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atis at iq-labs.net
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| Posted: Wed Jan 16, 2008 8:10 am Post subject: [asterisk-users] bad sound quality after Redirect |
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On 1/16/08, Franz Schwartau <franz at electromail.org> wrote:
| Quote: | 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 |
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