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 < [email protected]> 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:* [email protected] [mailto: > [email protected]] *För *Anthony Minessale > *Skickat:* den 13 april 2009 20:38 > *Till:* [email protected] > *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 < > [email protected]> 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" <[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 > > _______________________________________________ > 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 > > > > > _______________________________________________ > 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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:[email protected] <msn%[email protected]> > GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:[email protected] <sip%[email protected]> > iax:[email protected]/888 > googletalk:[email protected]<googletalk%3aconf%[email protected]> > pstn:213-799-1400 > !DSPAM:49e3899632939315582408! > > _______________________________________________ > 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[email protected] <msn%[email protected]> GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[email protected] <sip%[email protected]> iax:[email protected]/888 googletalk:[email protected]<googletalk%3aconf%[email protected]> pstn:213-799-1400
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