[Asterisk-Users] slight echo via sip provider

[Florian Overkamp]

Hi,

Damon Estep wrote:
> Here is the setup; analog phone <> Linksys ata <> asterisk <> sip 
> provider sonus GSX 9000 <> PSTN <> called party.
> 
> The caller on the analog phone connected to the ATA hears no echo at all.
> 
> The called party has a slight echo of their voice.
> 
> All of the Zapata.conf echotraining, echocancel, etc do not seem to 
> apply here as there is no zap channel involved in the call.

Correct.

> I assume that since the echo is toward the called party who is on the 
> other side of the provider sonus softswitch and somewhere on the PSTN, 
> that the echo is really coming from the providers media gateway/softswitch.

This is possible, but not really likely. Most decent service providers 
use digital equipment and would (should) not introduct additional echo 
on their end.

However, it is very well possible that your Linksys ATA and the 
connected analog phone are causing the echo. I'm not sure about the 
capabilities of the Linksys, but with Sipura's you can modify the line 
impedance settings to best match your equipment.

Look for the Regional Tab at the top. There is a setting called FXS Port 
Impedance. Try various options in there - they should match your phone.


Best regards,
Florian