[asterisk-users] multicastRTp

Johann Steinwendtner steinwendtner at gmx.net
Sat Aug 9 12:27:34 CDT 2014

On 2014-08-08 21:54, Jerry Geis wrote:
> On Thu, Aug 7, 2014 at 2:53 PM, Jerry Geis <geisj at pagestation.com <mailto: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...
> If I call using the dial plan everything seems to work...
> Is there an issue with using call files ?????
> Channel: MulticastRTP/basic/ <>
> 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?

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