Re: [Openbts-discuss] No Audio between calls

[Arthur R. <art...@gm...>]

Hi Kostas,

I only have callerID issues when i call a ms from a softphone. When I check
the logs, it reeks of an asterisk misconfiguration (softphone in default
context and ms in openbts context, links fails with callerid), but I do not
have enough knowledge to correct it. Will study later.

I have the correct callerid when I call from ms to ms.
I use the latest asterisk package in ubuntu (i believe 14.x)

And maybe there is a codec issue cause echo calls do not drop, will look
into it. Are you using A5/3 ? I am, will try without it to check for
differences.

Regards,

Arthur

Le Tue Feb 03 2015 at 10:14:39, Kostas Tsalikis <k_t...@wi...>
a écrit :

 I don't think it's a clock issue, maybe an asterisk issue. The clock gives
> stability so the MSs can attach to the OpenBTS network.
>
> Anyway, in a call inside the OpenBTS network, between two MSs, I have no
> audio and the caller id is 0000000.
>
> There shouldn't be a NAT issue, because there is no NAT(bridged adapter
> between USRP N210 and the Ubuntu VM), and I haven't changed any of the
> initial configurations on sip or extentions.conf.
>
> Does anyone know if there are codec issues with Asterisk 11.7.0.4?
>
> Thanks,
> Kostas Tsalikis
>
>  On 2/2/2015 1:00 μμ, Arthur Rabaté wrote:
>
> Hi Kostas,
>
> I also have this issue. I received my B200 on friday, still waiting for
> the gpsdo.
>
>  The echo call always works, the tone call worked at the very beginning
> but cannot be completed now.
>
>  Calls between phones are working 10% of the time, 90% are without sound.
> I believed the issue was linked to the lack of a good clock, but will look
> in the logs.
>
>  Also, that's for the Asterisk geeks out there :
>  - I can't call a softphone; I can call my ms from a softphone, yet the
> displayed caller id on the gsm is 0000000 (and I do not get audio neither,
> gpsdo issue ?)
> - But an MS to Softphone initiated call fails everytime. maybe I should
> configure my extensions to route calls conveniently, but it does not
> correct the callerid-on-the-gsm issue.
>
>
> Le Mon Feb 02 2015 at 11:44:21, Kostas Tsalikis <k_t...@wi...>
> a écrit :
>
>>  Hi everyone,
>>
>> I have installed OpenBTS 4.0 with Asterisk 11.7.0.4 with Ubuntu 12.04(in
>> vmware fusion). In the vm, I have one network adapter, bridged. I can make
>> calls between the phones, using the default extension and sip
>> configurations, but there is no sound between the calls. When the call is
>> answered, I get from asterisk:
>>
>>         probation passed - setting RTP source address to 127.0.0.1:16576
>>
>> I don't know if is a codec problem with asterisk.
>>
>> Has anyone any suggestions?
>>
>> Thanks,
>> Kostas Tsalikis
>>
>>
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>