On Thu, Aug 7, 2014 at 2:53 PM, Jerry Geis <ge...@pagestation.com> wrote:
> I am using a cyberdata "sip paging adapter" and with the > Dial(MulticastRTP/basic/IP:port) and with > tshark I see the RTP data, my device looks like its accepting the data > and I hear a click for my relay on my device so it would seem its > accepting the call, > however - I hear no audio... > > Asterisk 11.11.0 is what I am using. > What might be wrong here? > Thanks, > > jerry > If I call using the dial plan everything seems to work... Is there an issue with using call files ????? Channel: MulticastRTP/basic/239.168.3.10:11000 It all seems to work, I see multicast audio, the unit answers, I just get no audio or crappy audio... Is the codec not set right in that case from a call file? How do I set the codec for multicastrtp in a call file? might make sense that speak live the codec is already established but from a call file there is no codec.... Any thoughts or how do I set the codec in a call file for multicast to try it? Thanks, Jerry
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