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<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Yes, I’m starting to realize that... :) but you to get
everything right – if I want to bridge a call, using the dialplan, then the
only way is to use loopback, right? If I don’t want a loopback I’m able to
bridge to the destination directly?<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>//Peter<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Från:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] <b>För </b>Anthony
Minessale<br>
<b>Skickat:</b> den 14 april 2009 17:27<br>
<b>Till:</b> freeswitch-users@lists.freeswitch.org<br>
<b>Ämne:</b> Re: [Freeswitch-users] Use of loopback channels and bridge() in
scripts...<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal style='margin-bottom:12.0pt'>yes,<br>
<br>
But if you plan is to bridge the call, the loopback channel is completely
unnecessary.<br>
Be careful how much control you want =D getting a phone call up and running is
more work<br>
than you think (see switch_ivr_originate.c)<br>
<br>
<o:p></o:p></p>
<div>
<p class=MsoNormal>On Tue, Apr 14, 2009 at 8:24 AM, Peter Olsson <<a
href="mailto:peter.olsson@visionutveckling.se">peter.olsson@visionutveckling.se</a>>
wrote:<o:p></o:p></p>
<div>
<div>
<p><span style='font-size:10.0pt;color:#1F497D'>Anthony,</span><o:p></o:p></p>
<p><span style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p><span style='font-size:10.0pt;color:#1F497D'>Yes, it seems to work correct
now. I did a couple of test calls, and tha audio was good – thanks!</span><o:p></o:p></p>
<p><span style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p><span style='font-size:10.0pt;color:#1F497D'>Another question about this
scenario...</span><o:p></o:p></p>
<p><span style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p><span style='font-size:10.0pt;color:#1F497D'>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?</span><o:p></o:p></p>
<p><span style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p><span style='font-size:10.0pt;color:#1F497D'>Regards,</span><o:p></o:p></p>
<p><span style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
<p><span style='font-size:10.0pt;color:#1F497D'>Peter</span><o:p></o:p></p>
<p><span style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p>
<div style='border:none;border-top:solid windowtext 1.0pt;padding:3.0pt 0cm 0cm 0cm;
border-color:-moz-use-text-color -moz-use-text-color'>
<p><b><span style='font-size:10.0pt'>Från:</span></b><span style='font-size:
10.0pt'> <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org"
target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a
href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>]
<b>För </b>Anthony Minessale<br>
<b>Skickat:</b> den 13 april 2009 20:38<br>
<b>Till:</b> <a href="mailto:freeswitch-users@lists.freeswitch.org"
target="_blank">freeswitch-users@lists.freeswitch.org</a><br>
<b>Ämne:</b> Re: [Freeswitch-users] Use of loopback channels and bridge() in
scripts...</span><o:p></o:p></p>
</div>
<div>
<div>
<p> <o:p></o:p></p>
<p style='margin-bottom:12.0pt'>see how it works in latest trunk 13011<br>
<br>
nontheless you can just say <br>
<br>
session.execute("bridge", "loopback/5000");<br>
<br>
and get the same result without touching that other channel.<br>
<br>
when the call fails, you will have an originate_disposition variable in session
you can check.<br>
<br>
<o:p></o:p></p>
<div>
<p>On Mon, Apr 13, 2009 at 11:21 AM, Peter Olsson <<a
href="mailto:peter.olsson@visionutveckling.se" target="_blank">peter.olsson@visionutveckling.se</a>>
wrote:<o:p></o:p></p>
<p> 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..<br>
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.<br>
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?<br>
<br>
---<br>
<br>
if (session.ready()) {<br>
<br>
session.answer();<br>
<br>
new_session = new Session("loopback/5000", session);<br>
new_session.waitForAnswer();<br>
<br>
bridge(session, new_session);<br>
<br>
// Not sure if this is needed - I've tried with it both enabled and
disabled<br>
session.hangup();<br>
new_session.hangup();<br>
}<br>
<span style='color:#888888'><br>
Peter</span><o:p></o:p></p>
<div>
<div>
<p><br>
<br>
On 09-04-13 17.54, "Anthony Minessale" <<a
href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>>
wrote:<br>
<br>
1) When you say latest, which rev does that mean? we change revs pretty often.<br>
2) Do you have a minimal script that reproduces your issue.<br>
3) is there a reason you cannot just session.execute("bridge", dest);<br>
instead of doing it manually (which is a process not for the faint
at heart)?<br>
<br>
<br>
<br>
On Mon, Apr 13, 2009 at 10:29 AM, Peter Olsson <<a
href="mailto:peter.olsson@visionutveckling.se" target="_blank">peter.olsson@visionutveckling.se</a>>
wrote:<br>
I have two problems that I haven't been able to solve. I've done the same tests
in both javascript, and in .NET.<br>
<br>
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.<br>
<br>
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.<br>
<br>
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.<br>
<br>
I'm using the latest SVN trunk, and my OS is Windows XP.<br>
<br>
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.<br>
<br>
Has anyone else experienced this?<br>
<br>
Regards,<br>
<br>
Peter Olsson<br>
<br>
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-- <br>
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<br>
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-- <br>
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<br>
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