Anthony,
Yes, it seems to work correct now. I did a couple of test calls,
and tha audio was good – thanks!
Another question about this scenario...
When doing a session.transfer(”5000”), this will
transfer the call directly into the dialplan without the use of
loopback-channels. But that way it’s not possible to do it in a controlled
way. Shouldn’t it be possible to do the same thing with a bridge? As soon
as the call is bridged, it gets ”rid of” unneccecary loopback
channels, and connecting the two endpoints directly – cause by then it
should be two ”normal” endpoints talking?
Regards,
Peter
Från:
freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] För Anthony
Minessale
Skickat: den 13 april 2009 20:38
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Use of loopback channels and bridge() in
scripts...
see how it works in latest
trunk 13011
nontheless you can just say
session.execute("bridge", "loopback/5000");
and get the same result without touching that other channel.
when the call fails, you will have an originate_disposition variable in session
you can check.
On Mon, Apr 13, 2009 at 11:21 AM, Peter Olsson <peter.olsson@visionutveckling.se>
wrote:
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" <anthony.minessale@gmail.com>
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 <peter.olsson@visionutveckling.se>
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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