Re: [Openbts-discuss] No Audio between calls

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

I also am using VMware Fusion, but as everything is handled inside the VM
(Asterisk/oopenBTS/Transceiver), I got it working even without a virtual
ethernet adapter.
So I do not think NAT is our problem.

I'll try and install userv 12.04 on a laptop with usb3 this weekend, may be
able to report on monday.

Until then, if anyone has a clue...

Thanks,

Arthur

Le Fri Feb 13 2015 at 15:21:05, Kostas Tsalikis <k_t...@wi...>
a écrit :

>  Hello to list and Arthur,
>
> the last two weeks I didn't manage to work with OpenBTS at all. Arthur,
> there is no surprise that GPSDO didn't help, it only helps MS finding and
> camping to the OpenBTS network.
> I think that the problem with no sound between phone calls that me and
> Arthur are experiencing is a NAT problem and not a clock issue, please list
> give any thoughts on it.
>
> Arthur I think I'm using VMware fusion, don't remember which version, the
> host is a mac laptop. For ciphering I use the default, I don't have OpenBTS
> available at the time, so I can't answer for sure, but again, I think that
> our problem has to do something with NAT.
>
> The strange part from my behalf is that I have a bridged switch on the vm,
> so as far as i know, there shouldn't be any nat problem, but yet again,
> something may be slipping from my grasp.
>
> Thanks,
> Kostas Tsalikis
>
>
> On 13/2/2015 10:16 πμ, Arthur Rabaté wrote:
>
> 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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