asterisk users - Apr 2008 - Problems with Quality Voice in a Asterisk-E1-Unicall

I Have with Asterisk- Unicall - E1 (MFC/R2).

Days before a install a Digium Card TE122P with hardware echo 
cancelation, these because a had a echo in some in and out calls.

I replaced the card.   I no more echo but in my conversation the voice 
start to doing things.  Like after a minutes i start hearing the voice 
cut. or the cant hearme..

I remove in the zaptel.conf the echotraining.    I dont know if i really 
need to do these changes in the unicall.conf.???

In my Asterisk am using GXP2020 Grandstream what is better ulaw,alaw,g729???

I apreciate any help.

Thanks

Ruben

The E1 use ALAW, if you want to avoid trans-coding use ALAW in your
phones as well. In any call you have 2 call legs, callee and caller,
try to isolate the problem and determine if the audio is really coming
that bad from the E1, you can use ztmonitor to hook into the E1 and
listen to the audio. If the audio you get using ztmonitor is deficient
then you know it has nothing to do with trans-coding or the codec you
use in your phones.

Is the Digium card missing interrupts? (zttool will tell you so)

Mois?s Silva

On Sun, Apr 20, 2008 at 3:23 PM, Ruben Zamora <ruben.zamora at zys.com.mx>
wrote:
> I Have with Asterisk- Unicall - E1 (MFC/R2).
>
>  Days before a install a Digium Card TE122P with hardware echo
>  cancelation, these because a had a echo in some in and out calls.
>
>  I replaced the card.   I no more echo but in my conversation the voice
>  start to doing things.  Like after a minutes i start hearing the voice
>  cut. or the cant hearme..
>
>  I remove in the zaptel.conf the echotraining.    I dont know if i really
>  need to do these changes in the unicall.conf.???
>
>  In my Asterisk am using GXP2020 Grandstream what is better
ulaw,alaw,g729???
>
>  I apreciate any help.
>
>  Thanks
>
>  Ruben
>
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-- 
"I do not agree with what you have to say, but I'll defend to the
death your right to say it." Voltaire

Hello guys,

I thought it would be neat if we had a SIP client for Asterisk working in Adobe
Flash, but as far as I know, Flash only supports TCP. I know that Asterisk (at
least v1.6) can handle SIP communication over TCP, but I was wondering is there
a possibility to route audio stream over TCP too?

Regards,
Alex