Re: [Openbts-discuss] No Audio/RTP in incoming calls

[Heiko R. <hr...@py...>]

Hello Ralph,

good to know...
VMware Workstation or ESX(i) ?

Heiko
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On 3/1/2016 9:14 AM, Ralph A. Schmid, dk5ras wrote:
>
> Huh?! At least with vmware this all works just fine.
>
> Ralph.
>
> *From:*Heiko Rehm [mailto:hr...@py...]
> *Sent:* Tuesday, March 1, 2016 05:31
> *To:* Daigle, Frédéric; ope...@li...
> *Subject:* Re: [Openbts-discuss] No Audio/RTP in incoming calls
>
> Hello Frederic,
>
> Indeed am on a Virtual Box Lab...
> Will try 'physical'
>
> Thanks for the Tip!
>
> Heiko
>
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> On 3/1/2016 1:02 AM, Daigle, Frédéric wrote:
>
>     Hi Heiko,
>
>     I had the same kind of problem when I was using Virtual Box virtual machine to run my server. When I switched to a physical server, the problem disappeared.
>
>     Frédéric
>
>     ________________________________________
>
>     De : Heiko Rehm<hr...@py...> <mailto:hr...@py...>
>
>     Envoyé : 29 février 2016 00:47
>
>     À :ope...@li...
>     <mailto:ope...@li...>
>
>     Objet : Re: [Openbts-discuss] No Audio/RTP in incoming calls
>
>     Hello Community,
>
>     am still hanging on the same issue below...
>
>     nobody has/had that issue below?
>
>     Heiko
>
>     On 2/26/2016 9:34 PM, Heiko Rehm wrote:
>
>         Hello Community,
>
>         after juggling quite a bit with openBTS and it's components, I got it
>
>         mostly working.
>
>         Right now, I am hanging on a no-Audio issue:
>
>         --> No Audio/RTP on Calls between Handsets.
>
>         --> No Audio on incoming calls from outside (through Asterisk) to a
>
>         Handset.
>
>         Upon such a Inter-MS call or on a incoming call, the MS will properly
>
>         ring and the called party can pick up the call.
>
>         Instead of any Audio - total silence. After a few seconds, the call
>
>         get dropped (I guess thats some normal silence detection watchdog)
>
>         Outgoing calls to and through the Asterisk are working fine inclusive
>
>         two-way Audio.
>
>         SMS's are working fine too.
>
>         Codecs are all set to 'gsm' on the Asterisk side.
>
>         Any idea what could be causing this issue?
>
>         Thanks,
>
>         Heiko
>
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