VoIP Mailing List Archives

[gk at exram.de]

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@3c.co.uk>
To: gk@exram.de