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.
<br><br>
<div><span class="gmail_quote">On 10/21/05, <b class="gmail_sendername">John Todd</b> <<a href="mailto:jtodd@loligo.com">jtodd@loligo.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">>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>_______________________________________________<br>Asterisk-Dev mailing list<br><a href="mailto:Asterisk-Dev@lists.digium.com">Asterisk-Dev@lists.digium.com</a><br><a href="http://lists.digium.com/mailman/listinfo/asterisk-dev">
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