> The echo between our extensions (using Polycom 550 handsets) disappears > once I removed the Digium echo module.
Are you routing internal calls from SIP -> DAHDI -> SIP? The digium echo module will not have any effect on pure SIP <-> SIP calls. Do you have acoustic echo cancellation active on the Polycom phones? > What kind of settings do you recommend for the "txgain and rxgain"? Ideally, you will need to measure to find out what settings you want. See this page on the wiki (see the note on values for PRI circuits): http://www.voip-info.org/wiki/view/Asterisk+zapata+gain+adjustment (use dahdi_monitor instead of ztmonitor) You can also just experiment with different values. Change just one setting at a time, and then reload Dahdi. Try this to start: txgain = 0.0 rxgain = 1.0 and then on the asterisk cli, enter: module reload chan_dahdi.so If that doesn't help, try increasing to rxgain=2.0. Keep going until it sounds better. You may want to try negative values for txgain. > Do I > make the gain changes in chan_dahdi.conf? Yes. Make sure to put them before your channel numbers. You can specify values on a per-channel basis. > This is my system.conf: > bchan=1-23 > dchan=24 > echocanceller=mg2,1-23 Did you use these same settings when you were using the hardware echo module? - Noah _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
