Re: [Openbts-discuss] No Audio between calls

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

Hi,

So I still have a "no-sound" issue in my calls. When I check the logs I get
nothing helping : I have some stale bursts from the board, but I also have
them when my calls are completed.

So far I have checked those :
Power and RXgain does not change anything (well, within reasonable limits),
MS TX neither.
Ciphering (A5/3) does not change anything
Asterisk works (if not, none of my calls would work, would they ?).
Happens a lot more in MS to MS calls, sometimes in MS to Echo or other
tests.

I am using a VM, on a standard - core i5 - mac mini.
The last thing I did not check is whereas I will still have these issues
running this on a bare-metal install.

Anyone here have been encountering those issues ?

Thanks,

Arthur

Le Wed Feb 11 2015 at 19:48:17, Arthur Rabaté <art...@gm...> a
écrit :

> (Oh, so yesterday did not last for long ...)
> Le mar. 10 févr. 2015 à 22:59, Arthur Rabaté <art...@gm...> a
> écrit :
>
> Hi Kostas,
>>
>> So, I finally cracked it. Received the GPSDO today and that did not
>> change anything...
>>
>> I had MS RSSI at -70, increased it to -50. Also increased power from -15
>> to -5. All voice calls are handled very well now.
>>
>> Le Tue Feb 03 2015 at 11:54:29, 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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>>>>
>>>>