[asterisk-users] Simplex voice on TDM410P

[Tzafrir Cohen]

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
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