[Asterisk-Dev] Help needed for custom application

[Jonathan k. Creasy]

So your application, fdial, would work sort of like a monitor except with 2 way audio. It could be used say for a call center where  a supervisor wanted to listen to a call and provide coaching to the agent but not have the supervisor's audio go to the person on the other end of the call. 

 

Is this correct?

 

-Jonathan

 

________________________________

From: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of BJ Weschke
Sent: Saturday, October 22, 2005 3:41 PM
To: Asterisk Developers Mailing List
Subject: Re: [Asterisk-Dev] Help needed for custom application

 

 What you're requesting to be done here is going to require not only some of the functionality from app_dial to establish a call, but you will also need MeetMe itself or some of the internals from MeetMe to mux the RTP frames out to two destinations instead of the one original destination. 

On 10/21/05, John Todd <jtodd at loligo.com> wrote: 

>Hi folks,
>I've been tasked with writing a very special application for Asterisk which is
>needed for a very special project (I have no idea whether I may give out 
>details, so I'll stay on the safe side and won't). Since I still have
>difficulties in understanding the Asterisk internals I thought I could try
>asking the people who probably know the most about Asterisk :-) 
>
>Just so you don't get me wrong, I don't want you to develop that application
>for me, I just need some hints at how to do it.
>
>The application I need to develop (let's call it fdial) can be described as a 
>forking dial application (which only needs to support SIP). It should act
>almost like a normal dial at first, but when a special control frame arrives
>from a channel it should dial to a second address, and when the call is 
>established the voice streams should be sent to both destinations (and vice
>versa, but not between the two destinations. So the voice stream is forking
>to two destinations. One of the established destination channels may then be 
>hung up (obviously at least one destination channel must be up, otherwise we
>have a normal hangup).
>
>I know it sounds weird :-)
>
>I've looked into app_dial.c, but that code is doing way too much to easily 
>understand what is necessary to do a call... and especially having two active
>destinations channel at a time is something that no other applications seems
>to do, not even MeetMe (MeetMe seems to do conferencing/voice stream mixing 
>via a Zaptel device, doesn't it ?).
>
>Has anyone some suggestions for me ? Is it even possible without really nasty
>hacks to Asterisk itself ? Does anyone know of an application that does
>something remotely like the fdial I need to implement ? 
>
>Thanks for your help in advance,
>       Marc Haisenko
>--
>Marc Haisenko
>Linux Solutions
>Be O.K. service group GmbH
>
>Rüdesheimer Straße 7
>D-80686 München 
>Tel:   +49 (0)89 - 548 43 33 21
>Fax:   +49 (0)89 - 548 43 33 29
>e-mail: haisenko at be-ok.com
>http://www.be-ok.com


This sounds like you're trying to do an
intercept, but with dual way audio?  I would
suggest looking at app_chanspy for ideas.  I'm
somewhat unclear on what the actual goal is of
your project, so perhaps some diagrams might help 
here.

JT


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