[asterisk-users] multicastRTp

Jerry Geis geisj at pagestation.com
Fri Aug 8 14:54:32 CDT 2014


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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140808/2b9ae02d/attachment.html>


More information about the asterisk-users mailing list