[asterisk-dev] request for integration of new dialplan application into trunk - howto proceed?

Andreas Brodmann andreas.brodmann at gmail.com
Sat Jan 19 04:53:30 CST 2008

2008/1/12, Johansson Olle E <oej at edvina.net>:
> 12 jan 2008 kl. 15.25 skrev Andreas Brodmann:
> > Hi,
> >
> > I have written a dialplan application which supports
> > the multicast rtp feature of snom and linksys phones
> > and other devices (like barix.com) which accept
> > multicast or unicast rtp streams.
> >
> > I'd be glad if this would be accepted as part of the
> > default asterisk distribution.
> >
> > Can anyone tell me how I have to proceed?
> Great!
> Go to bugs.digium.com and register for an account and follow the
> instructions on how to accept the license
> (you do that on line now). Read the instructions carefully!
> Then open a bug report, upload the source code and we'll help you
> forward. Find us in #asterisk-dev on the IRC freenode.net server
> to get help. We'll be happy to help you.
> The source code has to work with Asterisk svn trunk.

Hi Olle,

I have made sure the application compiles against trunk (had to slightly
modify the call to ast_config_load)
and have now uploaded the source (app_rtpstream) as well as a sample config

The issue/bug Id is 11797.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20080119/4e743fcc/attachment.htm 

More information about the asterisk-dev mailing list