[asterisk-users] Simplex voice on TDM410P

[Andres]

jonas kellens wrote:

> 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.

I have seen this happen with the aggressive echo cancel algorithms.  You 
might want to look into that.

Andres
http://www.neuroredes.com

>
> 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
>>    
>>
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