2008/1/12, Johansson Olle E <<a href="mailto:oej@edvina.net">oej@edvina.net</a>>:<div><span class="gmail_quote"></span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>12 jan 2008 kl. 15.25 skrev Andreas Brodmann:<br><br>> Hi,<br>><br>> I have written a dialplan application which supports<br>> the multicast rtp feature of snom and linksys phones<br>> and other devices (like
<a href="http://barix.com">barix.com</a>) which accept<br>> multicast or unicast rtp streams.<br>><br>> I'd be glad if this would be accepted as part of the<br>> default asterisk distribution.<br>><br>> Can anyone tell me how I have to proceed?
<br><br>Great!<br><br>Go to <a href="http://bugs.digium.com">bugs.digium.com</a> and register for an account and follow the<br>instructions on how to accept the license<br>(you do that on line now). Read the instructions carefully!
<br><br>Then open a bug report, upload the source code and we'll help you<br>forward. Find us in #asterisk-dev on the IRC <a href="http://freenode.net">freenode.net</a> server<br>to get help. We'll be happy to help you.
<br><br>The source code has to work with Asterisk svn trunk.</blockquote><div><br>Hi Olle,<br><br>I have made sure the application compiles against trunk (had to slightly modify the call to ast_config_load)<br>and have now uploaded the source (app_rtpstream) as well as a sample config file.
<br><br>The issue/bug Id is 11797.<br><br>Regards,<br>Andreas<br></div><br></div>