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geisj at pagestation.com
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 Posted: Thu Aug 07, 2014 1:53 pm Post subject: [asterisk-users] multicastRTp |
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I am using a cyberdata "sip paging adapter" and with the Dial(MulticastRTP/basic/IP:port) and withtshark 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 |
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geisj at pagestation.com
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| Posted: Fri Aug 08, 2014 2:55 pm Post subject: [asterisk-users] multicastRTp |
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On Thu, Aug 7, 2014 at 2:53 PM, Jerry Geis <geisj@pagestation.com (geisj@pagestation.com)> wrote:
| Quote: | 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
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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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steinwendtner at gmx.net
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| Posted: Sat Aug 09, 2014 12:27 pm Post subject: [asterisk-users] multicastRTp |
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On 2014-08-08 21:54, Jerry Geis wrote:
| Quote: |
On Thu, Aug 7, 2014 at 2:53 PM, Jerry Geis <geisj@pagestation.com <mailto:geisj@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/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?
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Please check this link and see if this applies to you:
http://www.voip-info.org/wiki/view/Asterisk+MulticastRTP+channels
Regards
Hans
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