1. The latest trunk I've tried with is 13008. Since I'm not doing anything
for production yet (just testing/evaluating), so I tend to update as soon as
there is new version available..
2. Yep, you will find it below. In javascript - my sample for .NET does
basically the same thing, with the same result, except that it also won't drop
the loopback-a call leg.
3. Hmm.. Not really - I'm just in the middle of learning FS, so I guess I'm
not 100% sure what I'm doing.. :) What I want to be able to do is to dial into
a script, let the script dial another extension, and bridge them together when
the other party answers the call. I also need to take care of call setup
problems - if the other part doesn't respond, is unavailable or busy in the
phone - so I though this was the only way? If I use the
session.execute("bridge"..), will I be able to control the call if it couldn't
be connected?
---
if (session.ready()) {
session.answer();
new_session = new Session("loopback/5000", session);
new_session.waitForAnswer();
bridge(session, new_session);
// Not sure if this is needed - I've tried with it both enabled and disabled
session.hangup();
new_session.hangup();
}
Peter
On 09-04-13 17.54, "Anthony Minessale" <[email protected]> wrote:
1) When you say latest, which rev does that mean? we change revs pretty often.
2) Do you have a minimal script that reproduces your issue.
3) is there a reason you cannot just session.execute("bridge", dest);
instead of doing it manually (which is a process not for the faint at
heart)?
On Mon, Apr 13, 2009 at 10:29 AM, Peter Olsson
<[email protected]> wrote:
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
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