Re: [Openbts-discuss] No Audio between calls

[Ralph A. S. d. <ra...@sc...>]

Interesting. I have the "no audio" calls sometimes directly after starting OpenBTS, with the first call. The following calls are just fine, so I decided to live with it. I am using VMware. 

 

Ralph.

 

From: Arthur Rabaté [mailto:art...@gm...] 
Sent: Saturday, February 14, 2015 11:47
To: Kostas Tsalikis; OpenBTS Discuss
Subject: Re: [Openbts-discuss] No Audio between calls

 

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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