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Re: [Openbts-discuss] No Audio between calls
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From: Kostas T. <k_t...@wi...> - 2015-02-13 14:21:10
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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... > <mailto:art...@gm...>> a écrit : > > (Oh, so yesterday did not last for long ...) > Le mar. 10 févr. 2015 à 22:59, Arthur Rabaté > <art...@gm... <mailto: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... <mailto: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... > <mailto: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... >> <mailto: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 <http://127.0.0.1:16576> >> >> I don't know if is a codec problem with asterisk. >> >> Has anyone any suggestions? >> >> Thanks, >> Kostas Tsalikis >> >> >> ------------------------------------------------------------------------------ >> Dive into the World of Parallel Programming. The >> Go Parallel Website, >> sponsored by Intel and developed in partnership >> with Slashdot Media, is your >> hub for all things parallel software development, >> from weekly thought >> leadership blogs to news, videos, case studies, >> tutorials and more. 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