[Asterisk-Users] slight echo via sip provider

[Damon Estep]

When we make calls out of asterisk to the PSTN via a SIP termination
service provider the called party gets a slight echo of their voice.

 

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.

 

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.

 

Is there anything that can be done in asterisk to mitigate this, or is
this purely an issue that must be resolved on the providers sonus
switch?

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