Re: [Openbts-discuss] No Audio between calls

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

I have installed a new openbts system on bare metal laptop.
I have not seen any issue since, except that I still can not get gprs and
dtmf working.

I guess our issues were related to the use of VMware Fusion.

Regards,

Arthur

Le Fri Feb 13 2015 at 15:30:47, Arthur Rabaté <art...@gm...> a
écrit :

> 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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