Re: [Openbts-discuss] No Audio between calls

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

(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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>>>
>>>