Hello,
I am working on a trixbox based system with a TDM410P connected to 3
phone lines from the CO. The asterisk box is on a full duplex 100Mb LAN
with some polycom and Aastra SIP phones. In general everything works.
the problem I am trying to solve is that if both parties to a call speak
at the same time one of the voices gets cut out such that the talker A
cannot hear what talker B is saying. When talker A stops talking, he/she
can then hear what talker B says. This issue occurs across all the
different phones we have set up. I have played with the OSLEC settings
in the thoughts that the echo cancellation was being a bit ambitious, to
no avail. Any recommendations on how to best troubleshoot / correct this
issue?
Thanks and Regards,
Nate
I have posted a similar problem earlier on this mailing list with my Asterisk-system + TDM410 + Grandstream telephones. But there has not yet been a response to this. My client is also experiencing a 'simplex' conversation. There seems that audio can only flow 1 one way at the same time. What I have tried is change the codec on the internal SIP-network from alaw to gsm (so more compression, less bandwidth needed) but problem not yet resolved. Also I don't know where to begin to look for the problem... So, I'm curious for the solution. Greetingz, Jonas. On Sat, 2009-05-30 at 14:35 -0400, Nathanial A. Byrnes wrote:> Hello, > I am working on a trixbox based system with a TDM410P connected to 3 > phone lines from the CO. The asterisk box is on a full duplex 100Mb LAN > with some polycom and Aastra SIP phones. In general everything works. > the problem I am trying to solve is that if both parties to a call speak > at the same time one of the voices gets cut out such that the talker A > cannot hear what talker B is saying. When talker A stops talking, he/she > can then hear what talker B says. This issue occurs across all the > different phones we have set up. I have played with the OSLEC settings > in the thoughts that the echo cancellation was being a bit ambitious, to > no avail. Any recommendations on how to best troubleshoot / correct this > issue? > > Thanks and Regards, > Nate-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090530/a6faed0e/attachment.htm
On Sat, May 30, 2009 at 02:35:43PM -0400, Nathanial A. Byrnes wrote:> Hello, > I am working on a trixbox based system with a TDM410P connected to 3 > phone lines from the CO. The asterisk box is on a full duplex 100Mb LAN > with some polycom and Aastra SIP phones. In general everything works. > the problem I am trying to solve is that if both parties to a call speak > at the same time one of the voices gets cut out such that the talker A > cannot hear what talker B is saying. When talker A stops talking, he/she > can then hear what talker B says. This issue occurs across all the > different phones we have set up. I have played with the OSLEC settings > in the thoughts that the echo cancellation was being a bit ambitious, to > no avail. Any recommendations on how to best troubleshoot / correct this > issue?Is there a problem with SIP<->SIP call? I suppose there isn't and that you've already tested that. You can try taking SIP out of the equasion: originate DAHDI/N/NUMBER application Playback demo-instruct Or: originate DAHDI/N/NUMBER application Echo for an echo test. Here 'N' is the DAHDI channel number to dial through and NUMBER is the number to dial. Another thing you can do is to use dahdi_monitor to either look at the audio levels or record the audio. You can clearly see there when there's no audio in a certain direction. This is the audio Asterisk sends to Zaptel and recieves from it. Note, however, that the digits that DAHDI dials ar esents as a DAHDI_DIAL ioctl rather than an explicit digit sound. What versions of Asteirsk and DAHDI are those? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.cohen at xorcom.com +972-50-7952406 mailto:tzafrir.cohen at xorcom.com http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir