I have two problems that I haven't been able to solve. I've done the same tests in both javascript, and in .NET.
The two scripts are pretty simple, they just answer an incomming call, creates a new session, wait for an answer on the second call leg, and then bridge the two channels together. In both cases everything works just fine, but the audio is distorted. The destination I'm calling is "loopback/5000" - the sample IVR application included in FreeSWITCH. I first thought it was a codec issue, but even after trying to switch to different codecs the problem was the same. It more sounds like it's a timestamping issue - the voice is not distorted enough to be a bad codec, but it reads way to fast (mayby twice the "normal" speed). When doing a direct transfer() to the other destination this works just fine, but I need to be able to have some extra logic to tell if the destination is available or not. The second problem occurs only in .NET. After doing this sample there is as loopback channel still hanging around. It seems like the call creates a loopback-a and loopback-b, the loopback-b dissapears as it should (when the call has been disconnected), but the other one stays there. When doing the same in javascript this doesn't seem to occur. I'm using the latest SVN trunk, and my OS is Windows XP. I found bug FSCORE-349 in Jira, which seems to point in to the direction that there might be a bug with the loopback channels in some cases, but I could not find anything about the audio which plays too fast. Has anyone else experienced this? Regards, Peter Olsson _______________________________________________ Freeswitch-users mailing list [email protected] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
