On 2014-08-08 21:54, Jerry Geis wrote:
On Thu, Aug 7, 2014 at 2:53 PM, Jerry Geis <[email protected] <mailto:[email protected]>> 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/239.168.3.10:11000 <http://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?
Please check this link and see if this applies to you: http://www.voip-info.org/wiki/view/Asterisk+MulticastRTP+channels Regards Hans -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
