[Freeswitch-users] Re-2: Re-2: FreeSwitch Complex IVR System

Fri Apr 17 06:37:53 PDT 2009

Hi Dave,

didn't fully get your explanation. Maybe I should explain how I do it right now.

I am using the http://www.codeplex.com/eventsocket library. I take my x-lite phone with number 1000 and call into FS with number 301234. Under this number I launch my app as async full.

<extension name="MyApp">
   <condition field="destination_number" expression="^30[0-9][0-9][0-9][0-9]$">
      <action application="sleep data="2000"/>
      <action application="socket" data=192.168.49.237:8021 async full"/>
   </condition>
</extension>

I my app I initiate the FSEventManager with the same IP address and to subscribe to all Events and with correct set password from event_socket.conf.xml on port 8021.

So now a call is coming in and I receive several Events. On Progress Event I check wether it is an inbound or outbound call direction and create a call Object in my app and set ANI and DNIS before I call a Method who takes care of the next steps. The call is then answered. After taht I get the next Event, who informs me about the call that has been answered. Then I play a greeting and when the greeting ends I get a new event informing me that the greeting has ended. Now comes the struggle

I send a RecordSession Command, send a StartDTMF command and after that I send a Bridge Command to bridge the call from my extension 1000 to 1001. At thta point I hear moh on ext 1000 and the ext 1001 rings. I can pickup and I can even answer and hanup or hangup before the bridge was fully established that all works fine even if I have found out thta this is only working because there are some predefinition ins defaut.conf.xml for these internal sip extensions.

Problem is that I need full control over the call and over the different call legs. Only the called party (1001) should be able to do something with DTMF input. If extension 1000 does dtmf it doesn't really matter.

This is just testing environment. Normally my FS box is behind a PBX as gateway to the ISDN Network. So all the predifined things for internal SIP phones won't work if I try it with phones attached to my PBX I think.

I am a little bit lost. I have much experience in phone apps but my programming skills are a little bit rusty, cause they are coming from the good old DOS times.

Any help or hints will be appreciated....thanks...Guido

-------- Original Message --------
Subject: Re: [Freeswitch-users] Re-2:  FreeSwitch Complex IVR System (16-Apr-2009 19:02)
From:    David Knell <dave at 3c.co.uk>
To:      gk at exram.de

> Hi Guido,
> 
> The event socket interface will give you DTMF events for bridged calls -
> just tried it and it works fine.  There's one mild snag, which is that
> outbound sockets (which are easier for inbound call handling) will only
> give you events relating to the specific call leg that's attached to
> that socket - i.e. you can use an outbound socket app to bridge that leg
> to an outbound call leg just fine, but you won't get events related to
> that outbound call.
> 
> So what we do is use an outbound socket app for call control and
> scripting, and have a separate inbound socket app which listens for call
> state changes and DTMF on all call legs, and a database table which
> glues the two together. 
> 
> Cheers --
> 
> Dave
> 
> 
> > Hi Dave,
> > 
> > thanks for the answer. I am playing around with FS and Event Socket Library 
> > for .NET. I get pretty much to run with this, but the reason why I came 
> > from Asterisk to FS is that I cannot get DTMF in a bridged call. I thought 
> > that I get an Event as soon one dtmf digit is recognized. Unfortunately 
> > this isn't the case.
> > 
> > If I use the default config files and map the keys with bind_meta_app the 
> > dtmf tones are recognized and the function behind the bound app is executed.
> >  Is this maybe a bug.
> > 
> > I have read about mod_managed and that I should use it, but I haven't found 
> > anything about the usage of it.
> > 
> > Any suggestions would help....
> > 
> > thanks...Guido
> > 
> > -------- Original Message --------
> > Subject: Re: [Freeswitch-users] FreeSwitch Complex IVR System (16-Apr-2009 
> > 17:35)
> > From:    David Knell <dave at 3c.co.uk>
> > To:      gk at exram.de
> > 
> > > Hi Guido,
> > > 
> > > My preferred way is to talk to FS through its event socket
> > > interface.  This allows you fully to control FS, whilst giving
> > > you the power to write the code in whatever language and on
> > > whatever platform you choose.
> > > 
> > > The documentation starts here:
> > > http://wiki.freeswitch.org/wiki/Mod_event_socket
> > > 
> > > Cheers --
> > > 
> > > Dave
> > > 
> > > > Hi @all
> > > >  
> > > > I have a question about a project I want to realize with FreeSwitch. I
> > > > want to do a complex IVR System which takes a call, do many things in
> > > > a MSSQL DB, send some Informations to one or many Middleware Servers
> > > > via TCP/IP, call one or more mobile phones, the first is able to take
> > > > the call, it can be that he must be able to hear a prompt before he is
> > > > actually connected to the first caller, then the conversation must be
> > > > recorded automatically and during the conversation it must be possible
> > > > for the called party to redirect the call by dtmf. I know that this is
> > > > all possible, but I want to know which way is the best to do all this?
> > > > 
> > > > 
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