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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>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. <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Is this correct?<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>-Jonathan<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
asterisk-dev-bounces@lists.digium.com
[mailto:asterisk-dev-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>BJ Weschke<br>
<b><span style='font-weight:bold'>Sent:</span></b> Saturday, October 22, 2005
3:41 PM<br>
<b><span style='font-weight:bold'>To:</span></b> Asterisk Developers Mailing
List<br>
<b><span style='font-weight:bold'>Subject:</span></b> Re: [Asterisk-Dev] Help
needed for custom application</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><font size=3
face="Times New Roman"><span style='font-size:12.0pt'> 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. <o:p></o:p></span></font></p>
<div>
<p class=MsoNormal><span class=gmailquote><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'>On 10/21/05, <b><span style='font-weight:bold'>John
Todd</span></b> <<a href="mailto:jtodd@loligo.com">jtodd@loligo.com</a>>
wrote:</span></font></span> <o:p></o:p></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>>Hi folks,<br>
>I've been tasked with writing a very special application for Asterisk which
is<br>
>needed for a very special project (I have no idea whether I may give out <br>
>details, so I'll stay on the safe side and won't). Since I still have<br>
>difficulties in understanding the Asterisk internals I thought I could try<br>
>asking the people who probably know the most about Asterisk :-) <br>
><br>
>Just so you don't get me wrong, I don't want you to develop that
application<br>
>for me, I just need some hints at how to do it.<br>
><br>
>The application I need to develop (let's call it fdial) can be described as
a <br>
>forking dial application (which only needs to support SIP). It should act<br>
>almost like a normal dial at first, but when a special control frame
arrives<br>
>from a channel it should dial to a second address, and when the call is <br>
>established the voice streams should be sent to both destinations (and vice<br>
>versa, but not between the two destinations. So the voice stream is forking<br>
>to two destinations. One of the established destination channels may then
be <br>
>hung up (obviously at least one destination channel must be up, otherwise
we<br>
>have a normal hangup).<br>
><br>
>I know it sounds weird :-)<br>
><br>
>I've looked into app_dial.c, but that code is doing way too much to easily <br>
>understand what is necessary to do a call... and especially having two
active<br>
>destinations channel at a time is something that no other applications
seems<br>
>to do, not even MeetMe (MeetMe seems to do conferencing/voice stream mixing
<br>
>via a Zaptel device, doesn't it ?).<br>
><br>
>Has anyone some suggestions for me ? Is it even possible without really
nasty<br>
>hacks to Asterisk itself ? Does anyone know of an application that does<br>
>something remotely like the fdial I need to implement ? <br>
><br>
>Thanks for your help in advance,<br>
> Marc Haisenko<br>
>--<br>
>Marc Haisenko<br>
>Linux Solutions<br>
>Be O.K. service group GmbH<br>
><br>
>Rüdesheimer Straße 7<br>
>D-80686 München <br>
>Tel: +49 (0)89 - 548 43 33 21<br>
>Fax: +49 (0)89 - 548 43 33 29<br>
>e-mail: <a href="mailto:haisenko@be-ok.com">haisenko@be-ok.com</a><br>
><a href="http://www.be-ok.com">http://www.be-ok.com</a><br>
<br>
<br>
This sounds like you're trying to do an<br>
intercept, but with dual way audio? I would<br>
suggest looking at app_chanspy for ideas. I'm<br>
somewhat unclear on what the actual goal is of<br>
your project, so perhaps some diagrams might help <br>
here.<br>
<br>
JT<br>
<br>
<br>
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