[Freeswitch-users] Use of loopback channels and bridge() in scripts...

Tue Apr 14 08:26:55 PDT 2009

yes,

But if you plan is to bridge the call, the loopback channel is completely
unnecessary.
Be careful how much control you want =D getting a phone call up and running
is more work
than you think (see switch_ivr_originate.c)

On Tue, Apr 14, 2009 at 8:24 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

>  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 at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *För *Anthony Minessale
> *Skickat:* den 13 april 2009 20:38
> *Till:* freeswitch-users at 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 at 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 at 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 at 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
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> !DSPAM:49e3899632939315582408!
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090414/e38e0519/attachment-0002.html 