On Thu, Aug 7, 2014 at 2:53 PM, Jerry Geis <geisj at 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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