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 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
