Question : asterisk: echo on SIP to FXO

I have a bunch of 7960 sip phones connected to a asterisk with regular lines connected via FXO modules to my phone company. Now people using the sip phones can hear themselfs while they`re talking but people on the other side cant... The latency is low and Im using the g711ulaw codec.

Should I decrease the gain? any other ways to cancel the echo?
Tom

Answer : asterisk: echo on SIP to FXO

Have a look at this page and the other links at the bottom. There are various causes and way to fix it depending on what hardware you have and this page covers the options.

http://www.voip-info.org/wiki-Asterisk+echo+analog+lines