Modify SIP Client Gain?

I have a new install of asterisk ( and am using Cisco 7940 as phones and am receiving a very slight echo. Sounds like delayed sidetone the internal person hears themselves and the other participants do not hear the echo. The PRI card I am using has a hardware echo canceller and I belive it is doing its job correctly. When I monitor a call using dahdi_monitor the TX side is off the chart with the 7940’s…if I use a soft phone, xlite in this case, the TX is around 50-75% and there is never any echo. If I change the TX gain in chan_dahdi you can audibly tell the difference but the level stays them same in dahdi_monitor. Any ideas? I am using these phones in other instillations without any issues. I am not seeing any gain settings on the 7940’s themselves…is there something in the asterisk config I can change?


What PRI card are you using?