On Sep 7, 2014 2:28 PM, "Joshua Colp" <jc...@digium.com> wrote: > > Johann Steinwendtner wrote: >> >> On 2014-09-07 17:07, Joshua Colp wrote: >>> >>> This is an automatically generated e-mail. To reply, visit: >>> https://reviewboard.asterisk.org/r/3981/ >>> >>> >> >>> Testing >>> >>> Originated a call to a UnicastRTP channel and sent it to a Playback. >>> Confirmed that RTP was sent to the provided IP address/port with the >>> given format. >>> >> >> Hello, can you please explain what you mean by "with the given format". >> There is a patch from John R. Covert which adds the capability of >> selecting the codec. Or is this not necessary in trunk. >> >> http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/48495 > > > The UnicastRTP dial string allows specifying the format. I did not touch MulticastRTP. > What does the dial string look like? I didn't see any documentation on it. Mind you I am using my phone for the code review.
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