Re: [Openbts-discuss] No Audio between calls

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

Hi Kostas,

Are you using a vm.

Still have an issue, 90% of my calls are either static or nothing. Happens
more with ms to ms calls than ms to softphone or echo.

May be a processing power issue ?
Le mar. 3 févr. 2015 à 11:54, Arthur Rabaté <art...@gm...> a
écrit :

> 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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>>