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,
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
